2 * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
3 * <2006> Lutz Mueller <lutz at topfrose dot de>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
18 * Boston, MA 02111-1307, USA.
21 * Unless otherwise indicated, Source Code is licensed under MIT license.
22 * See further explanation attached in License Statement (distributed in the file
25 * Permission is hereby granted, free of charge, to any person obtaining a copy of
26 * this software and associated documentation files (the "Software"), to deal in
27 * the Software without restriction, including without limitation the rights to
28 * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
29 * of the Software, and to permit persons to whom the Software is furnished to do
30 * so, subject to the following conditions:
32 * The above copyright notice and this permission notice shall be included in all
33 * copies or substantial portions of the Software.
35 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
36 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
37 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
38 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
39 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
40 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
44 * SECTION:element-rtspsrc
46 * Makes a connection to an RTSP server and read the data.
47 * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
48 * RealMedia/Quicktime/Microsoft extensions.
50 * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
51 * default rtspsrc will negotiate a connection in the following order:
52 * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
53 * protocols can be controlled with the #GstRTSPSrc:protocols property.
55 * rtspsrc currently understands SDP as the format of the session description.
56 * For each stream listed in the SDP a new rtp_stream%d pad will be created
57 * with caps derived from the SDP media description. This is a caps of mime type
58 * "application/x-rtp" that can be connected to any available RTP depayloader
61 * rtspsrc will internally instantiate an RTP session manager element
62 * that will handle the RTCP messages to and from the server, jitter removal,
63 * packet reordering along with providing a clock for the pipeline.
64 * This feature is implemented using the gstrtpbin element.
66 * rtspsrc acts like a live source and will therefore only generate data in the
70 * <title>Example launch line</title>
72 * gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
73 * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
77 * Last reviewed on 2006-08-18 (0.10.5)
86 #endif /* HAVE_UNISTD_H */
93 #include <gst/sdp/gstsdpmessage.h>
94 #include <gst/rtp/gstrtppayloads.h>
96 #include "gst/gst-i18n-plugin.h"
98 #include "gstrtspsrc.h"
101 #include <winsock2.h>
104 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
105 #define GST_CAT_DEFAULT (rtspsrc_debug)
107 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
110 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
112 /* templates used internally */
113 static GstStaticPadTemplate anysrctemplate =
114 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
117 GST_STATIC_CAPS_ANY);
119 static GstStaticPadTemplate anysinktemplate =
120 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
123 GST_STATIC_CAPS_ANY);
131 enum _GstRtspSrcRtcpSyncMode
138 enum _GstRtspSrcBufferMode
146 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
148 gst_rtsp_src_buffer_mode_get_type (void)
150 static GType buffer_mode_type = 0;
151 static const GEnumValue buffer_modes[] = {
152 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
153 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
154 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
155 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
159 if (!buffer_mode_type) {
161 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
163 return buffer_mode_type;
166 #define DEFAULT_LOCATION NULL
167 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
168 #define DEFAULT_DEBUG FALSE
169 #define DEFAULT_RETRY 20
170 #define DEFAULT_TIMEOUT 5000000
171 #define DEFAULT_UDP_BUFFER_SIZE 0x80000
172 #define DEFAULT_TCP_TIMEOUT 20000000
173 #define DEFAULT_LATENCY_MS 2000
174 #define DEFAULT_CONNECTION_SPEED 0
175 #define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
176 #define DEFAULT_DO_RTCP TRUE
177 #define DEFAULT_PROXY NULL
178 #define DEFAULT_RTP_BLOCKSIZE 0
179 #define DEFAULT_USER_ID NULL
180 #define DEFAULT_USER_PW NULL
181 #define DEFAULT_BUFFER_MODE BUFFER_MODE_AUTO
182 #define DEFAULT_PORT_RANGE NULL
183 #define DEFAULT_SHORT_HEADER FALSE
195 PROP_CONNECTION_SPEED,
204 PROP_UDP_BUFFER_SIZE,
209 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
211 gst_rtsp_nat_method_get_type (void)
213 static GType rtsp_nat_method_type = 0;
214 static const GEnumValue rtsp_nat_method[] = {
215 {GST_RTSP_NAT_NONE, "None", "none"},
216 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
220 if (!rtsp_nat_method_type) {
221 rtsp_nat_method_type =
222 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
224 return rtsp_nat_method_type;
227 static void gst_rtspsrc_finalize (GObject * object);
229 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
230 const GValue * value, GParamSpec * pspec);
231 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
232 GValue * value, GParamSpec * pspec);
234 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
235 gpointer iface_data);
237 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
240 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
241 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
243 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
245 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
246 GstStateChange transition);
247 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
248 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
250 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
251 GstRTSPMessage * response);
253 static void gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
255 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
256 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
258 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
259 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
261 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle,
263 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
264 gboolean only_close);
266 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
267 const gchar * uri, GError ** error);
269 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
270 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
271 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
272 GstRTSPStream * stream, GstEvent * event, gboolean source);
273 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event,
276 /* commands we send to out loop to notify it of events */
282 #define CMD_RECONNECT 5
285 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
287 gchar *__txt = _gst_element_error_printf text; \
288 gst_element_post_message (GST_ELEMENT_CAST (el), \
289 gst_message_new_progress (GST_OBJECT_CAST (el), \
290 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
294 /*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
295 #define gst_rtspsrc_parent_class parent_class
296 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
297 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
300 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
302 GObjectClass *gobject_class;
303 GstElementClass *gstelement_class;
304 GstBinClass *gstbin_class;
306 gobject_class = (GObjectClass *) klass;
307 gstelement_class = (GstElementClass *) klass;
308 gstbin_class = (GstBinClass *) klass;
310 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
312 gobject_class->set_property = gst_rtspsrc_set_property;
313 gobject_class->get_property = gst_rtspsrc_get_property;
315 gobject_class->finalize = gst_rtspsrc_finalize;
317 g_object_class_install_property (gobject_class, PROP_LOCATION,
318 g_param_spec_string ("location", "RTSP Location",
319 "Location of the RTSP url to read",
320 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
322 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
323 g_param_spec_flags ("protocols", "Protocols",
324 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
325 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
327 g_object_class_install_property (gobject_class, PROP_DEBUG,
328 g_param_spec_boolean ("debug", "Debug",
329 "Dump request and response messages to stdout",
330 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
332 g_object_class_install_property (gobject_class, PROP_RETRY,
333 g_param_spec_uint ("retry", "Retry",
334 "Max number of retries when allocating RTP ports.",
335 0, G_MAXUINT16, DEFAULT_RETRY,
336 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
338 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
339 g_param_spec_uint64 ("timeout", "Timeout",
340 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
341 0, G_MAXUINT64, DEFAULT_TIMEOUT,
342 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
344 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
345 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
346 "Fail after timeout microseconds on TCP connections (0 = disabled)",
347 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
348 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
350 g_object_class_install_property (gobject_class, PROP_LATENCY,
351 g_param_spec_uint ("latency", "Buffer latency in ms",
352 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
353 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
355 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
356 g_param_spec_uint64 ("connection-speed", "Connection Speed",
357 "Network connection speed in kbps (0 = unknown)",
358 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
359 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
361 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
362 g_param_spec_enum ("nat-method", "NAT Method",
363 "Method to use for traversing firewalls and NAT",
364 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
365 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
368 * GstRTSPSrc::do-rtcp
370 * Enable RTCP support. Some old server don't like RTCP and then this property
371 * needs to be set to FALSE.
375 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
376 g_param_spec_boolean ("do-rtcp", "Do RTCP",
377 "Send RTCP packets, disable for old incompatible server.",
378 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
383 * Set the proxy parameters. This has to be a string of the format
384 * [http://][user:passwd@]host[:port].
388 g_object_class_install_property (gobject_class, PROP_PROXY,
389 g_param_spec_string ("proxy", "Proxy",
390 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
391 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
394 * GstRTSPSrc::rtp_blocksize
396 * RTP package size to suggest to server.
400 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
401 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
402 "RTP package size to suggest to server (0 = disabled)",
403 0, 65536, DEFAULT_RTP_BLOCKSIZE,
404 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
406 g_object_class_install_property (gobject_class,
408 g_param_spec_string ("user-id", "user-id",
409 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
410 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
411 g_object_class_install_property (gobject_class, PROP_USER_PW,
412 g_param_spec_string ("user-pw", "user-pw",
413 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
414 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
417 * GstRTSPSrc::buffer-mode:
419 * Control the buffering and timestamping mode used by the jitterbuffer.
423 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
424 g_param_spec_enum ("buffer-mode", "Buffer Mode",
425 "Control the buffering algorithm in use",
426 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
427 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
430 * GstRTSPSrc::port-range:
432 * Configure the client port numbers that can be used to recieve RTP and
437 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
438 g_param_spec_string ("port-range", "Port range",
439 "Client port range that can be used to receive RTP and RTCP data, "
440 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
441 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
444 * GstRTSPSrc::udp-buffer-size:
446 * Size of the kernel UDP receive buffer in bytes.
450 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
451 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
452 "Size of the kernel UDP receive buffer in bytes, 0=default",
453 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
454 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
457 * GstRTSPSrc::short-header:
459 * Only send the basic RTSP headers for broken encoders.
463 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
464 g_param_spec_boolean ("short-header", "Short Header",
465 "Only send the basic RTSP headers for broken encoders",
466 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
468 gstelement_class->send_event = gst_rtspsrc_send_event;
469 gstelement_class->change_state = gst_rtspsrc_change_state;
471 gst_element_class_add_pad_template (gstelement_class,
472 gst_static_pad_template_get (&rtptemplate));
474 gst_element_class_set_details_simple (gstelement_class,
475 "RTSP packet receiver", "Source/Network",
476 "Receive data over the network via RTSP (RFC 2326)",
477 "Wim Taymans <wim@fluendo.com>, "
478 "Thijs Vermeir <thijs.vermeir@barco.com>, "
479 "Lutz Mueller <lutz@topfrose.de>");
481 gstbin_class->handle_message = gst_rtspsrc_handle_message;
483 gst_rtsp_ext_list_init ();
488 gst_rtspsrc_init (GstRTSPSrc * src)
493 if (WSAStartup (MAKEWORD (2, 2), &wsa_data) != 0) {
494 GST_ERROR_OBJECT (src, "WSAStartup failed: 0x%08x", WSAGetLastError ());
498 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
499 src->protocols = DEFAULT_PROTOCOLS;
500 src->debug = DEFAULT_DEBUG;
501 src->retry = DEFAULT_RETRY;
502 src->udp_timeout = DEFAULT_TIMEOUT;
503 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
504 src->latency = DEFAULT_LATENCY_MS;
505 src->connection_speed = DEFAULT_CONNECTION_SPEED;
506 src->nat_method = DEFAULT_NAT_METHOD;
507 src->do_rtcp = DEFAULT_DO_RTCP;
508 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
509 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
510 src->user_id = g_strdup (DEFAULT_USER_ID);
511 src->user_pw = g_strdup (DEFAULT_USER_PW);
512 src->buffer_mode = DEFAULT_BUFFER_MODE;
513 src->client_port_range.min = 0;
514 src->client_port_range.max = 0;
515 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
516 src->short_header = DEFAULT_SHORT_HEADER;
518 /* get a list of all extensions */
519 src->extensions = gst_rtsp_ext_list_get ();
521 /* connect to send signal */
522 gst_rtsp_ext_list_connect (src->extensions, "send",
523 (GCallback) gst_rtspsrc_send_cb, src);
525 /* protects the streaming thread in interleaved mode or the polling
526 * thread in UDP mode. */
527 src->stream_rec_lock = g_new (GStaticRecMutex, 1);
528 g_static_rec_mutex_init (src->stream_rec_lock);
530 /* protects our state changes from multiple invocations */
531 src->state_rec_lock = g_new (GStaticRecMutex, 1);
532 g_static_rec_mutex_init (src->state_rec_lock);
534 src->state = GST_RTSP_STATE_INVALID;
536 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
540 gst_rtspsrc_finalize (GObject * object)
544 rtspsrc = GST_RTSPSRC (object);
546 gst_rtsp_ext_list_free (rtspsrc->extensions);
547 g_free (rtspsrc->conninfo.location);
548 gst_rtsp_url_free (rtspsrc->conninfo.url);
549 g_free (rtspsrc->conninfo.url_str);
550 g_free (rtspsrc->user_id);
551 g_free (rtspsrc->user_pw);
554 gst_sdp_message_free (rtspsrc->sdp);
559 g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
560 g_free (rtspsrc->stream_rec_lock);
561 g_static_rec_mutex_free (rtspsrc->state_rec_lock);
562 g_free (rtspsrc->state_rec_lock);
568 G_OBJECT_CLASS (parent_class)->finalize (object);
571 /* a proxy string of the format [user:passwd@]host[:port] */
573 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
577 g_free (rtsp->proxy_user);
578 rtsp->proxy_user = NULL;
579 g_free (rtsp->proxy_passwd);
580 rtsp->proxy_passwd = NULL;
581 g_free (rtsp->proxy_host);
582 rtsp->proxy_host = NULL;
583 rtsp->proxy_port = 0;
590 /* we allow http:// in front but ignore it */
591 if (g_str_has_prefix (p, "http://"))
594 at = strchr (p, '@');
596 /* look for user:passwd */
597 col = strchr (proxy, ':');
598 if (col == NULL || col > at)
601 rtsp->proxy_user = g_strndup (p, col - p);
603 rtsp->proxy_passwd = g_strndup (col, at - col);
608 col = strchr (p, ':');
611 /* everything before the colon is the hostname */
612 rtsp->proxy_host = g_strndup (p, col - p);
614 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
616 rtsp->proxy_host = g_strdup (p);
617 rtsp->proxy_port = 8080;
623 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
625 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
626 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
629 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
631 rtspsrc->ptcp_timeout = NULL;
635 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
640 rtspsrc = GST_RTSPSRC (object);
644 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
645 g_value_get_string (value), NULL);
648 rtspsrc->protocols = g_value_get_flags (value);
651 rtspsrc->debug = g_value_get_boolean (value);
654 rtspsrc->retry = g_value_get_uint (value);
657 rtspsrc->udp_timeout = g_value_get_uint64 (value);
659 case PROP_TCP_TIMEOUT:
660 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
663 rtspsrc->latency = g_value_get_uint (value);
665 case PROP_CONNECTION_SPEED:
666 rtspsrc->connection_speed = g_value_get_uint64 (value);
668 case PROP_NAT_METHOD:
669 rtspsrc->nat_method = g_value_get_enum (value);
672 rtspsrc->do_rtcp = g_value_get_boolean (value);
675 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
677 case PROP_RTP_BLOCKSIZE:
678 rtspsrc->rtp_blocksize = g_value_get_uint (value);
681 if (rtspsrc->user_id)
682 g_free (rtspsrc->user_id);
683 rtspsrc->user_id = g_value_dup_string (value);
686 if (rtspsrc->user_pw)
687 g_free (rtspsrc->user_pw);
688 rtspsrc->user_pw = g_value_dup_string (value);
690 case PROP_BUFFER_MODE:
691 rtspsrc->buffer_mode = g_value_get_enum (value);
693 case PROP_PORT_RANGE:
697 str = g_value_get_string (value);
699 sscanf (str, "%u-%u",
700 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
702 rtspsrc->client_port_range.min = 0;
703 rtspsrc->client_port_range.max = 0;
707 case PROP_UDP_BUFFER_SIZE:
708 rtspsrc->udp_buffer_size = g_value_get_int (value);
710 case PROP_SHORT_HEADER:
711 rtspsrc->short_header = g_value_get_boolean (value);
714 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
720 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
725 rtspsrc = GST_RTSPSRC (object);
729 g_value_set_string (value, rtspsrc->conninfo.location);
732 g_value_set_flags (value, rtspsrc->protocols);
735 g_value_set_boolean (value, rtspsrc->debug);
738 g_value_set_uint (value, rtspsrc->retry);
741 g_value_set_uint64 (value, rtspsrc->udp_timeout);
743 case PROP_TCP_TIMEOUT:
747 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
748 rtspsrc->tcp_timeout.tv_usec;
749 g_value_set_uint64 (value, timeout);
753 g_value_set_uint (value, rtspsrc->latency);
755 case PROP_CONNECTION_SPEED:
756 g_value_set_uint64 (value, rtspsrc->connection_speed);
758 case PROP_NAT_METHOD:
759 g_value_set_enum (value, rtspsrc->nat_method);
762 g_value_set_boolean (value, rtspsrc->do_rtcp);
768 if (rtspsrc->proxy_host) {
770 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
774 g_value_take_string (value, str);
777 case PROP_RTP_BLOCKSIZE:
778 g_value_set_uint (value, rtspsrc->rtp_blocksize);
781 g_value_set_string (value, rtspsrc->user_id);
784 g_value_set_string (value, rtspsrc->user_pw);
786 case PROP_BUFFER_MODE:
787 g_value_set_enum (value, rtspsrc->buffer_mode);
789 case PROP_PORT_RANGE:
793 if (rtspsrc->client_port_range.min != 0) {
794 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
795 rtspsrc->client_port_range.max);
799 g_value_take_string (value, str);
802 case PROP_UDP_BUFFER_SIZE:
803 g_value_set_int (value, rtspsrc->udp_buffer_size);
805 case PROP_SHORT_HEADER:
806 g_value_set_boolean (value, rtspsrc->short_header);
809 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
815 find_stream_by_id (GstRTSPStream * stream, gint * id)
817 if (stream->id == *id)
824 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
826 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
833 find_stream_by_pt (GstRTSPStream * stream, gint * pt)
835 if (stream->pt == *pt)
842 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
844 GstElement *src = (GstElement *) a;
846 if (stream->udpsrc[0] == src)
848 if (stream->udpsrc[1] == src)
855 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
857 /* check qualified setup_url */
858 if (!strcmp (stream->conninfo.location, (gchar *) a))
860 /* check original control_url */
861 if (!strcmp (stream->control_url, (gchar *) a))
864 /* check if qualified setup_url ends with string */
865 if (g_str_has_suffix (stream->control_url, (gchar *) a))
871 static GstRTSPStream *
872 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
876 /* find and get stream */
877 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
878 return (GstRTSPStream *) lstream->data;
883 static const GstSDPBandwidth *
884 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
885 const GstSDPMedia * media, const gchar * type)
889 /* first look in the media specific section */
890 len = gst_sdp_media_bandwidths_len (media);
891 for (i = 0; i < len; i++) {
892 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
894 if (strcmp (bw->bwtype, type) == 0)
897 /* then look in the message specific section */
898 len = gst_sdp_message_bandwidths_len (sdp);
899 for (i = 0; i < len; i++) {
900 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
902 if (strcmp (bw->bwtype, type) == 0)
909 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
910 const GstSDPMedia * media, GstRTSPStream * stream)
912 const GstSDPBandwidth *bw;
914 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
915 stream->as_bandwidth = bw->bandwidth;
917 stream->as_bandwidth = -1;
919 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
920 stream->rr_bandwidth = bw->bandwidth;
922 stream->rr_bandwidth = -1;
924 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
925 stream->rs_bandwidth = bw->bandwidth;
927 stream->rs_bandwidth = -1;
931 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
932 const GstSDPConnection * conn)
934 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
937 if (conn->addrtype == NULL)
941 if (strcmp (conn->addrtype, "IP4") == 0)
942 stream->is_ipv6 = FALSE;
943 else if (strcmp (conn->addrtype, "IP6") == 0)
944 stream->is_ipv6 = TRUE;
949 g_free (stream->destination);
950 stream->destination = g_strdup (conn->address);
952 /* check for multicast */
953 stream->is_multicast =
954 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
956 stream->ttl = conn->ttl;
959 /* Go over the connections for a stream.
960 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
962 * - If we are dealing with a localhost address, we disable multicast
965 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
966 const GstSDPMedia * media, GstRTSPStream * stream)
968 const GstSDPConnection *conn;
971 /* first look in the media specific section */
972 len = gst_sdp_media_connections_len (media);
973 for (i = 0; i < len; i++) {
974 conn = gst_sdp_media_get_connection (media, i);
976 gst_rtspsrc_do_stream_connection (src, stream, conn);
978 /* then look in the message specific section */
979 if ((conn = gst_sdp_message_get_connection (sdp))) {
980 gst_rtspsrc_do_stream_connection (src, stream, conn);
984 static GstRTSPStream *
985 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
987 GstRTSPStream *stream;
988 const gchar *control_url;
989 const gchar *payload;
990 const GstSDPMedia *media;
992 /* get media, should not return NULL */
993 media = gst_sdp_message_get_media (sdp, idx);
997 stream = g_new0 (GstRTSPStream, 1);
998 stream->parent = src;
999 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
1001 stream->last_ret = GST_FLOW_NOT_LINKED;
1002 stream->added = FALSE;
1003 stream->disabled = FALSE;
1004 stream->id = src->numstreams++;
1005 stream->eos = FALSE;
1006 stream->discont = TRUE;
1007 stream->seqbase = -1;
1008 stream->timebase = -1;
1010 /* collect bandwidth information for this steam. FIXME, configure in the RTP
1011 * session manager to scale RTCP. */
1012 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
1014 /* collect connection info */
1015 gst_rtspsrc_collect_connections (src, sdp, media, stream);
1017 /* we must have a payload. No payload means we cannot create caps */
1018 /* FIXME, handle multiple formats. The problem here is that we just want to
1019 * take the first available format that we can handle but in order to do that
1020 * we need to scan for depayloader plugins. Scanning for payloader plugins is
1021 * also suboptimal because the user maybe just wants to save the raw stream
1022 * and then we don't care. */
1023 if ((payload = gst_sdp_media_get_format (media, 0))) {
1024 stream->pt = atoi (payload);
1026 stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
1028 GST_DEBUG ("mapping sdp session level attributes to caps");
1029 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, stream->caps);
1030 GST_DEBUG ("mapping sdp media level attributes to caps");
1031 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, stream->caps);
1033 if (stream->pt >= 96) {
1034 /* If we have a dynamic payload type, see if we have a stream with the
1035 * same payload number. If there is one, they are part of the same
1036 * container and we only need to add one pad. */
1037 if (find_stream (src, &stream->pt, (gpointer) find_stream_by_pt)) {
1038 stream->container = TRUE;
1039 GST_DEBUG ("found another stream with pt %d, marking as container",
1044 /* collect port number */
1045 stream->port = gst_sdp_media_get_port (media);
1047 /* get control url to construct the setup url. The setup url is used to
1048 * configure the transport of the stream and is used to identity the stream in
1049 * the RTP-Info header field returned from PLAY. */
1050 control_url = gst_sdp_media_get_attribute_val (media, "control");
1051 if (control_url == NULL)
1052 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1054 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1055 GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
1056 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1057 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1058 GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
1059 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1061 if (control_url != NULL) {
1062 stream->control_url = g_strdup (control_url);
1063 /* Build a fully qualified url using the content_base if any or by prefixing
1064 * the original request.
1065 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1066 * likely build a URL that the server will fail to understand, this is ok,
1067 * we will fail then. */
1068 if (g_str_has_prefix (control_url, "rtsp://"))
1069 stream->conninfo.location = g_strdup (control_url);
1074 if (g_strcmp0 (control_url, "*") == 0)
1078 base = src->control;
1079 else if (src->content_base)
1080 base = src->content_base;
1081 else if (src->conninfo.url_str)
1082 base = src->conninfo.url_str;
1086 /* check if the base ends or control starts with / */
1087 has_slash = g_str_has_prefix (control_url, "/");
1088 has_slash = has_slash || g_str_has_suffix (base, "/");
1090 /* concatenate the two strings, insert / when not present */
1091 stream->conninfo.location =
1092 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1095 GST_DEBUG_OBJECT (src, " setup: %s",
1096 GST_STR_NULL (stream->conninfo.location));
1098 /* we keep track of all streams */
1099 src->streams = g_list_append (src->streams, stream);
1107 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1111 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1114 gst_caps_unref (stream->caps);
1116 g_free (stream->destination);
1117 g_free (stream->control_url);
1118 g_free (stream->conninfo.location);
1120 for (i = 0; i < 2; i++) {
1121 if (stream->udpsrc[i]) {
1122 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1123 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1124 gst_object_unref (stream->udpsrc[i]);
1125 stream->udpsrc[i] = NULL;
1127 if (stream->channelpad[i]) {
1128 gst_object_unref (stream->channelpad[i]);
1129 stream->channelpad[i] = NULL;
1131 if (stream->udpsink[i]) {
1132 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1133 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1134 gst_object_unref (stream->udpsink[i]);
1135 stream->udpsink[i] = NULL;
1138 if (stream->fakesrc) {
1139 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1140 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1141 gst_object_unref (stream->fakesrc);
1142 stream->fakesrc = NULL;
1144 if (stream->srcpad) {
1145 gst_pad_set_active (stream->srcpad, FALSE);
1146 if (stream->added) {
1147 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1148 stream->added = FALSE;
1150 stream->srcpad = NULL;
1152 if (stream->rtcppad) {
1153 gst_object_unref (stream->rtcppad);
1154 stream->rtcppad = NULL;
1156 if (stream->session) {
1157 g_object_unref (stream->session);
1158 stream->session = NULL;
1164 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1168 GST_DEBUG_OBJECT (src, "cleanup");
1170 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1171 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1173 gst_rtspsrc_stream_free (src, stream);
1175 g_list_free (src->streams);
1176 src->streams = NULL;
1178 if (src->manager_sig_id) {
1179 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1180 src->manager_sig_id = 0;
1182 gst_element_set_state (src->manager, GST_STATE_NULL);
1183 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1184 src->manager = NULL;
1186 src->numstreams = 0;
1188 gst_structure_free (src->props);
1191 g_free (src->content_base);
1192 src->content_base = NULL;
1194 g_free (src->control);
1195 src->control = NULL;
1198 gst_rtsp_range_free (src->range);
1201 /* don't clear the SDP when it was used in the url */
1202 if (src->sdp && !src->from_sdp) {
1203 gst_sdp_message_free (src->sdp);
1208 #define PARSE_INT(p, del, res) \
1211 p = strstr (p, del); \
1221 #define PARSE_STRING(p, del, res) \
1224 p = strstr (p, del); \
1236 #define SKIP_SPACES(p) \
1237 while (*p && g_ascii_isspace (*p)) \
1242 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1245 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1246 gint * rate, gchar ** params)
1250 p = (gchar *) rtpmap;
1252 PARSE_INT (p, " ", *payload);
1260 PARSE_STRING (p, "/", *name);
1261 if (*name == NULL) {
1262 GST_DEBUG ("no rate, name %s", p);
1263 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1264 * streams seem to omit the rate. */
1271 p = strstr (p, "/");
1289 * Mapping SDP attributes to caps
1291 * prepend 'a-' to IANA registered sdp attributes names
1292 * (ie: not prefixed with 'x-') in order to avoid
1293 * collision with gstreamer standard caps properties names
1296 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1298 if (attributes->len > 0) {
1302 s = gst_caps_get_structure (caps, 0);
1304 for (i = 0; i < attributes->len; i++) {
1305 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1306 gchar *tofree, *key;
1310 /* skip some of the attribute we already handle */
1311 if (!strcmp (key, "fmtp"))
1313 if (!strcmp (key, "rtpmap"))
1315 if (!strcmp (key, "control"))
1317 if (!strcmp (key, "range"))
1320 /* string must be valid UTF8 */
1321 if (!g_utf8_validate (attr->value, -1, NULL))
1324 if (!g_str_has_prefix (key, "x-"))
1325 tofree = key = g_strdup_printf ("a-%s", key);
1329 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1330 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1337 * Mapping of caps to and from SDP fields:
1339 * m=<media> <UDP port> RTP/AVP <payload>
1340 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1341 * a=fmtp:<payload> <param>[=<value>];...
1344 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1347 const gchar *rtpmap;
1351 gchar *params = NULL;
1357 /* get and parse rtpmap */
1358 if ((rtpmap = gst_sdp_media_get_attribute_val (media, "rtpmap"))) {
1359 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
1361 if (payload != pt) {
1362 /* we ignore the rtpmap if the payload type is different. */
1363 g_warning ("rtpmap of wrong payload type, ignoring");
1369 /* if we failed to parse the rtpmap for a dynamic payload type, we have an
1373 /* else we can ignore */
1374 g_warning ("error parsing rtpmap, ignoring");
1377 /* dynamic payloads need rtpmap or we fail */
1381 /* check if we have a rate, if not, we need to look up the rate from the
1382 * default rates based on the payload types. */
1384 const GstRTPPayloadInfo *info;
1386 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
1387 /* dynamic types, use media and encoding_name */
1388 tmp = g_ascii_strdown (media->media, -1);
1389 info = gst_rtp_payload_info_for_name (tmp, name);
1392 /* static types, use payload type */
1393 info = gst_rtp_payload_info_for_pt (pt);
1397 if ((rate = info->clock_rate) == 0)
1400 /* we fail if we cannot find one */
1405 tmp = g_ascii_strdown (media->media, -1);
1406 caps = gst_caps_new_simple ("application/x-unknown",
1407 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
1409 s = gst_caps_get_structure (caps, 0);
1411 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
1413 /* encoding name must be upper case */
1415 tmp = g_ascii_strup (name, -1);
1416 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
1420 /* params must be lower case */
1421 if (params != NULL) {
1422 tmp = g_ascii_strdown (params, -1);
1423 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
1427 /* parse optional fmtp: field */
1428 if ((fmtp = gst_sdp_media_get_attribute_val (media, "fmtp"))) {
1434 /* p is now of the format <payload> <param>[=<value>];... */
1435 PARSE_INT (p, " ", payload);
1436 if (payload != -1 && payload == pt) {
1440 /* <param>[=<value>] are separated with ';' */
1441 pairs = g_strsplit (p, ";", 0);
1442 for (i = 0; pairs[i]; i++) {
1444 const gchar *val, *key;
1446 /* the key may not have a '=', the value can have other '='s */
1447 valpos = strstr (pairs[i], "=");
1449 /* we have a '=' and thus a value, remove the '=' with \0 */
1451 /* value is everything between '=' and ';'. We split the pairs at ;
1452 * boundaries so we can take the remainder of the value. Some servers
1453 * put spaces around the value which we strip off here. Alternatively
1454 * we could strip those spaces in the depayloaders should these spaces
1455 * actually carry any meaning in the future. */
1456 val = g_strstrip (valpos + 1);
1458 /* simple <param>;.. is translated into <param>=1;... */
1461 /* strip the key of spaces, convert key to lowercase but not the value. */
1462 key = g_strstrip (pairs[i]);
1463 if (strlen (key) > 1) {
1464 tmp = g_ascii_strdown (key, -1);
1465 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
1477 g_warning ("rtpmap type not given for dynamic payload %d", pt);
1482 g_warning ("rate unknown for payload type %d", pt);
1488 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
1489 gint * rtpport, gint * rtcpport)
1492 GstStateChangeReturn ret;
1493 GstElement *udpsrc0, *udpsrc1;
1494 gint tmp_rtp, tmp_rtcp;
1498 src = stream->parent;
1504 /* Start at next port */
1505 tmp_rtp = src->next_port_num;
1507 if (stream->is_ipv6)
1508 host = "udp://[::0]";
1510 host = "udp://0.0.0.0";
1512 /* try to allocate 2 UDP ports, the RTP port should be an even
1513 * number and the RTCP port should be the next (uneven) port */
1516 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
1517 tmp_rtp >= src->client_port_range.max)
1520 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
1521 if (udpsrc0 == NULL)
1522 goto no_udp_protocol;
1523 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
1525 if (src->udp_buffer_size != 0)
1526 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
1529 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
1530 if (ret == GST_STATE_CHANGE_FAILURE) {
1532 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
1535 if (++count > src->retry)
1538 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1539 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1540 gst_object_unref (udpsrc0);
1542 GST_DEBUG_OBJECT (src, "retry %d", count);
1545 goto no_udp_protocol;
1548 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
1549 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
1551 /* check if port is even */
1552 if ((tmp_rtp & 0x01) != 0) {
1553 /* port not even, close and allocate another */
1554 if (++count > src->retry)
1557 GST_DEBUG_OBJECT (src, "RTP port not even");
1559 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1560 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1561 gst_object_unref (udpsrc0);
1563 GST_DEBUG_OBJECT (src, "retry %d", count);
1568 /* allocate port+1 for RTCP now */
1569 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
1570 if (udpsrc1 == NULL)
1571 goto no_udp_rtcp_protocol;
1574 tmp_rtcp = tmp_rtp + 1;
1575 if (src->client_port_range.max > 0 && tmp_rtcp >= src->client_port_range.max)
1578 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
1580 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
1581 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
1582 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
1583 if (ret == GST_STATE_CHANGE_FAILURE) {
1584 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
1586 if (++count > src->retry)
1589 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
1590 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1591 gst_object_unref (udpsrc0);
1593 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
1594 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1595 gst_object_unref (udpsrc1);
1599 GST_DEBUG_OBJECT (src, "retry %d", count);
1603 /* all fine, do port check */
1604 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
1605 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
1607 /* this should not happen... */
1608 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
1611 /* we keep these elements, we configure all in configure_transport when the
1612 * server told us to really use the UDP ports. */
1613 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
1614 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
1616 /* keep track of next available port number when we have a range
1618 if (src->next_port_num != 0)
1619 src->next_port_num = tmp_rtcp + 1;
1626 GST_DEBUG_OBJECT (src, "could not get UDP source");
1631 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
1635 no_udp_rtcp_protocol:
1637 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
1642 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
1643 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
1649 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1650 gst_object_unref (udpsrc0);
1653 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1654 gst_object_unref (udpsrc1);
1661 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
1668 GstClockTime base_time = GST_CLOCK_TIME_NONE;
1671 event = gst_event_new_flush_start ();
1672 GST_DEBUG_OBJECT (src, "start flush");
1674 state = GST_STATE_PAUSED;
1676 event = gst_event_new_flush_stop (TRUE);
1677 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
1680 state = GST_STATE_PLAYING;
1682 state = GST_STATE_PAUSED;
1683 clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
1685 base_time = gst_clock_get_time (clock);
1686 gst_object_unref (clock);
1689 gst_rtspsrc_push_event (src, event, FALSE);
1690 gst_rtspsrc_loop_send_cmd (src, cmd, flush);
1692 /* set up manager before data-flow resumes */
1693 /* to manage jitterbuffer buffer mode */
1695 gst_element_set_base_time (GST_ELEMENT_CAST (src->manager), base_time);
1696 /* and to have base_time trickle further down,
1697 * e.g. to jitterbuffer for its timeout handling */
1698 if (base_time != -1)
1699 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
1702 /* make running time start start at 0 again */
1703 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1704 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1706 for (i = 0; i < 2; i++) {
1708 if (stream->udpsrc[i]) {
1709 if (base_time != -1)
1710 gst_element_set_base_time (stream->udpsrc[i], base_time);
1711 gst_element_set_state (stream->udpsrc[i], state);
1715 /* for tcp interleaved case */
1716 if (base_time != -1)
1717 gst_element_set_base_time (GST_ELEMENT_CAST (src), base_time);
1720 static GstRTSPResult
1721 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
1722 GstRTSPMessage * message, GTimeVal * timeout)
1727 ret = gst_rtsp_connection_send (conn, message, timeout);
1729 ret = GST_RTSP_ERROR;
1734 static GstRTSPResult
1735 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
1736 GstRTSPMessage * message, GTimeVal * timeout)
1741 ret = gst_rtsp_connection_receive (conn, message, timeout);
1743 ret = GST_RTSP_ERROR;
1749 gst_rtspsrc_get_position (GstRTSPSrc * src)
1754 query = gst_query_new_position (GST_FORMAT_TIME);
1755 /* should be known somewhere down the stream (e.g. jitterbuffer) */
1756 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1757 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1761 if (stream->srcpad) {
1762 if (gst_pad_query (stream->srcpad, query)) {
1763 gst_query_parse_position (query, &fmt, &pos);
1764 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
1765 GST_TIME_ARGS (pos));
1766 src->last_pos = pos;
1776 gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
1778 src->state = GST_RTSP_STATE_SEEKING;
1779 /* PLAY will add the range header now. */
1780 src->need_range = TRUE;
1786 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
1791 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
1793 gboolean flush, skip;
1796 GstSegment seeksegment = { 0, };
1800 GST_DEBUG_OBJECT (src, "doing seek with event");
1802 gst_event_parse_seek (event, &rate, &format, &flags,
1803 &cur_type, &cur, &stop_type, &stop);
1805 /* no negative rates yet */
1809 /* we need TIME format */
1810 if (format != src->segment.format)
1813 GST_DEBUG_OBJECT (src, "doing seek without event");
1815 cur_type = GST_SEEK_TYPE_SET;
1816 stop_type = GST_SEEK_TYPE_SET;
1819 /* get flush flag */
1820 flush = flags & GST_SEEK_FLAG_FLUSH;
1821 skip = flags & GST_SEEK_FLAG_SKIP;
1823 /* now we need to make sure the streaming thread is stopped. We do this by
1824 * either sending a FLUSH_START event downstream which will cause the
1825 * streaming thread to stop with a WRONG_STATE.
1826 * For a non-flushing seek we simply pause the task, which will happen as soon
1827 * as it completes one iteration (and thus might block when the sink is
1828 * blocking in preroll). */
1830 GST_DEBUG_OBJECT (src, "starting flush");
1831 gst_rtspsrc_flush (src, TRUE, FALSE);
1834 gst_task_pause (src->task);
1838 /* we should now be able to grab the streaming thread because we stopped it
1839 * with the above flush/pause code */
1840 GST_RTSP_STREAM_LOCK (src);
1842 GST_DEBUG_OBJECT (src, "stopped streaming");
1844 /* copy segment, we need this because we still need the old
1845 * segment when we close the current segment. */
1846 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
1848 /* configure the seek parameters in the seeksegment. We will then have the
1849 * right values in the segment to perform the seek */
1851 GST_DEBUG_OBJECT (src, "configuring seek");
1852 gst_segment_do_seek (&seeksegment, rate, format, flags,
1853 cur_type, cur, stop_type, stop, &update);
1856 /* figure out the last position we need to play. If it's configured (stop !=
1857 * -1), use that, else we play until the total duration of the file */
1858 if ((stop = seeksegment.stop) == -1)
1859 stop = seeksegment.duration;
1861 playing = (src->state == GST_RTSP_STATE_PLAYING);
1863 /* if we were playing, pause first */
1865 /* obtain current position in case seek fails */
1866 gst_rtspsrc_get_position (src);
1867 gst_rtspsrc_pause (src, FALSE, FALSE);
1870 gst_rtspsrc_do_seek (src, &seeksegment);
1872 /* and continue playing */
1874 gst_rtspsrc_play (src, &seeksegment, FALSE);
1876 /* prepare for streaming again */
1878 /* if we started flush, we stop now */
1879 GST_DEBUG_OBJECT (src, "stopping flush");
1880 gst_rtspsrc_flush (src, FALSE, playing);
1883 /* now we did the seek and can activate the new segment values */
1884 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
1886 /* if we're doing a segment seek, post a SEGMENT_START message */
1887 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
1888 gst_element_post_message (GST_ELEMENT_CAST (src),
1889 gst_message_new_segment_start (GST_OBJECT_CAST (src),
1890 src->segment.format, src->segment.position));
1893 /* now create the newsegment */
1894 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
1895 " to %" G_GINT64_FORMAT, src->segment.position, stop);
1897 /* store the newsegment event so it can be sent from the streaming thread. */
1898 if (src->start_segment)
1899 gst_event_unref (src->start_segment);
1900 src->start_segment = gst_event_new_segment (&src->segment);
1903 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
1904 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1905 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1906 stream->discont = TRUE;
1910 GST_RTSP_STREAM_UNLOCK (src);
1917 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
1922 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
1928 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
1932 gboolean res = TRUE;
1935 src = GST_RTSPSRC_CAST (parent);
1937 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
1938 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
1940 switch (GST_EVENT_TYPE (event)) {
1941 case GST_EVENT_SEEK:
1942 res = gst_rtspsrc_perform_seek (src, event);
1946 case GST_EVENT_NAVIGATION:
1947 case GST_EVENT_LATENCY:
1955 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
1956 res = gst_pad_send_event (target, event);
1957 gst_object_unref (target);
1959 gst_event_unref (event);
1962 gst_event_unref (event);
1968 /* this is the final event function we receive on the internal source pad when
1969 * we deal with TCP connections */
1971 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
1977 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
1979 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
1980 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
1982 switch (GST_EVENT_TYPE (event)) {
1983 case GST_EVENT_SEEK:
1985 case GST_EVENT_NAVIGATION:
1986 case GST_EVENT_LATENCY:
1988 gst_event_unref (event);
1995 /* this is the final query function we receive on the internal source pad when
1996 * we deal with TCP connections */
1998 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2002 gboolean res = TRUE;
2004 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2006 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2007 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2009 switch (GST_QUERY_TYPE (query)) {
2010 case GST_QUERY_POSITION:
2015 case GST_QUERY_DURATION:
2019 gst_query_parse_duration (query, &format, NULL);
2022 case GST_FORMAT_TIME:
2023 gst_query_set_duration (query, format, src->segment.duration);
2031 case GST_QUERY_LATENCY:
2033 /* we are live with a min latency of 0 and unlimited max latency, this
2034 * result will be updated by the session manager if there is any. */
2035 gst_query_set_latency (query, TRUE, 0, -1);
2045 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2047 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2051 gboolean res = FALSE;
2053 src = GST_RTSPSRC_CAST (parent);
2055 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2056 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2058 switch (GST_QUERY_TYPE (query)) {
2059 case GST_QUERY_DURATION:
2063 gst_query_parse_duration (query, &format, NULL);
2066 case GST_FORMAT_TIME:
2067 gst_query_set_duration (query, format, src->segment.duration);
2075 case GST_QUERY_SEEKING:
2079 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2080 if (format == GST_FORMAT_TIME) {
2082 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2084 /* seeking without duration is unlikely */
2085 seekable = seekable && src->seekable && src->segment.duration &&
2086 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2088 /* FIXME ?? should we have 0 and segment.duration here; see demuxers */
2089 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
2090 src->segment.start, src->segment.stop);
2097 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2099 /* forward the query to the proxy target pad */
2101 res = gst_pad_query (target, query);
2102 gst_object_unref (target);
2111 /* callback for RTCP messages to be sent to the server when operating in TCP
2113 static GstFlowReturn
2114 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2117 GstRTSPStream *stream;
2118 GstFlowReturn res = GST_FLOW_OK;
2123 GstRTSPMessage message = { 0 };
2124 GstRTSPConnection *conn;
2126 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2127 src = stream->parent;
2129 data = gst_buffer_map (buffer, &bsize, NULL, GST_MAP_READ);
2132 gst_rtsp_message_init_data (&message, stream->channel[1]);
2134 /* lend the body data to the message */
2135 gst_rtsp_message_take_body (&message, data, size);
2137 if (stream->conninfo.connection)
2138 conn = stream->conninfo.connection;
2140 conn = src->conninfo.connection;
2142 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2143 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2144 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2146 /* and steal it away again because we will free it when unreffing the
2148 gst_rtsp_message_steal_body (&message, &data, &size);
2149 gst_rtsp_message_unset (&message);
2151 gst_buffer_unmap (buffer, data, size);
2152 gst_buffer_unref (buffer);
2157 static GstPadProbeReturn
2158 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2160 GstRTSPSrc *src = user_data;
2162 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2163 GST_DEBUG_PAD_NAME (pad));
2165 /* activate the streams */
2166 GST_OBJECT_LOCK (src);
2167 if (!src->need_activate)
2170 src->need_activate = FALSE;
2171 GST_OBJECT_UNLOCK (src);
2173 gst_rtspsrc_activate_streams (src);
2175 return GST_PAD_PROBE_OK;
2179 GST_OBJECT_UNLOCK (src);
2180 return GST_PAD_PROBE_OK;
2184 /* this callback is called when the session manager generated a new src pad with
2185 * payloaded RTP packets. We simply ghost the pad here. */
2187 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2190 GstPadTemplate *template;
2193 GstRTSPStream *stream;
2196 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2198 GST_RTSP_STATE_LOCK (src);
2200 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2201 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2202 goto unknown_stream;
2204 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %d, PT %d", id, ssrc, pt);
2206 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2208 goto unknown_stream;
2210 /* create a new pad we will use to stream to */
2211 template = gst_static_pad_template_get (&rtptemplate);
2212 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2213 gst_object_unref (template);
2216 stream->added = TRUE;
2217 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2218 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2219 gst_pad_set_active (stream->srcpad, TRUE);
2220 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2222 /* check if we added all streams */
2224 for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
2225 stream = (GstRTSPStream *) lstream->data;
2227 GST_DEBUG_OBJECT (src, "stream %p, container %d, disabled %d, added %d",
2228 stream, stream->container, stream->disabled, stream->added);
2230 /* a container stream only needs one pad added. Also disabled streams don't
2232 if (!stream->container && !stream->disabled && !stream->added) {
2237 GST_RTSP_STATE_UNLOCK (src);
2240 GST_DEBUG_OBJECT (src, "We added all streams");
2241 /* when we get here, all stream are added and we can fire the no-more-pads
2243 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2251 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2252 GST_RTSP_STATE_UNLOCK (src);
2259 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2261 GstRTSPStream *stream;
2264 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2266 GST_RTSP_STATE_LOCK (src);
2267 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2269 goto unknown_stream;
2271 caps = stream->caps;
2273 gst_caps_ref (caps);
2274 GST_RTSP_STATE_UNLOCK (src);
2280 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2281 GST_RTSP_STATE_UNLOCK (src);
2287 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2289 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2295 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos (), TRUE);
2301 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2307 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2309 GstRTSPSrc *src = stream->parent;
2311 GST_DEBUG_OBJECT (src, "source in session %u received BYE", stream->id);
2313 gst_rtspsrc_do_stream_eos (src, stream);
2317 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2319 GstRTSPSrc *src = stream->parent;
2321 GST_DEBUG_OBJECT (src, "source in session %u timed out", stream->id);
2323 gst_rtspsrc_do_stream_eos (src, stream);
2327 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
2329 GstRTSPStream *stream;
2331 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
2333 /* get stream for session */
2334 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2336 gst_rtspsrc_do_stream_eos (src, stream);
2341 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
2343 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
2347 /* try to get and configure a manager */
2349 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
2350 GstRTSPTransport * transport)
2352 const gchar *manager;
2354 GstStateChangeReturn ret;
2356 /* find a manager */
2357 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
2361 GST_DEBUG_OBJECT (src, "using manager %s", manager);
2363 /* configure the manager */
2364 if (src->manager == NULL) {
2365 GObjectClass *klass;
2368 if (!(src->manager = gst_element_factory_make (manager, NULL))) {
2370 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
2374 goto use_no_manager;
2376 if (!(src->manager = gst_element_factory_make (manager, NULL)))
2377 goto manager_failed;
2380 /* we manage this element */
2381 gst_bin_add (GST_BIN_CAST (src), src->manager);
2383 GST_OBJECT_LOCK (src);
2384 target = GST_STATE_TARGET (src);
2385 GST_OBJECT_UNLOCK (src);
2387 ret = gst_element_set_state (src->manager, target);
2388 if (ret == GST_STATE_CHANGE_FAILURE)
2389 goto start_manager_failure;
2391 g_object_set (src->manager, "latency", src->latency, NULL);
2393 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
2394 if (g_object_class_find_property (klass, "buffer-mode")) {
2395 if (src->buffer_mode != BUFFER_MODE_AUTO) {
2396 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
2398 gboolean need_slave;
2400 const gchar *encoding;
2402 /* buffer mode pauses are handled by adding offsets to buffer times,
2403 * but some depayloaders may have a hard time syncing output times
2404 * with such input times, e.g. container ones, most notably ASF */
2405 /* TODO alternatives are having an event that indicates these shifts,
2406 * or having rtsp extensions provide suggestion on buffer mode */
2407 need_slave = stream->container;
2408 if (stream->caps && (s = gst_caps_get_structure (stream->caps, 0)) &&
2409 (encoding = gst_structure_get_string (s, "encoding-name")))
2410 need_slave = need_slave || (strcmp (encoding, "X-ASF-PF") == 0);
2411 GST_DEBUG_OBJECT (src, "auto buffering mode, need_slave %d",
2413 /* valid duration implies not likely live pipeline,
2414 * so slaving in jitterbuffer does not make much sense
2415 * (and might mess things up due to bursts) */
2416 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
2417 src->segment.duration && !need_slave) {
2418 GST_DEBUG_OBJECT (src, "selected buffer");
2419 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER,
2422 GST_DEBUG_OBJECT (src, "selected slave");
2423 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
2428 /* connect to signals if we did not already do so */
2429 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
2431 src->manager_sig_id =
2432 g_signal_connect (src->manager, "pad-added",
2433 (GCallback) new_manager_pad, src);
2434 src->manager_ptmap_id =
2435 g_signal_connect (src->manager, "request-pt-map",
2436 (GCallback) request_pt_map, src);
2438 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
2442 /* we stream directly to the manager, get some pads. Each RTSP stream goes
2443 * into a separate RTP session. */
2444 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
2445 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
2447 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
2448 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
2451 /* now configure the bandwidth in the manager */
2452 if (g_signal_lookup ("get-internal-session",
2453 G_OBJECT_TYPE (src->manager)) != 0) {
2454 GObject *rtpsession;
2456 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
2459 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
2461 stream->session = rtpsession;
2463 if (stream->as_bandwidth != -1) {
2464 GST_INFO_OBJECT (src, "setting AS: %f",
2465 (gdouble) (stream->as_bandwidth * 1000));
2466 g_object_set (rtpsession, "bandwidth",
2467 (gdouble) (stream->as_bandwidth * 1000), NULL);
2469 if (stream->rr_bandwidth != -1) {
2470 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
2471 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
2474 if (stream->rs_bandwidth != -1) {
2475 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
2476 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
2479 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2481 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
2483 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
2485 g_signal_connect (rtpsession, "on-ssrc-active",
2486 (GCallback) on_ssrc_active, stream);
2497 GST_DEBUG_OBJECT (src, "cannot get a session manager");
2502 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
2505 start_manager_failure:
2507 GST_DEBUG_OBJECT (src, "could not start session manager");
2512 /* free the UDP sources allocated when negotiating a transport.
2513 * This function is called when the server negotiated to a transport where the
2514 * UDP sources are not needed anymore, such as TCP or multicast. */
2516 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
2520 for (i = 0; i < 2; i++) {
2521 if (stream->udpsrc[i]) {
2522 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
2523 gst_object_unref (stream->udpsrc[i]);
2524 stream->udpsrc[i] = NULL;
2529 /* for TCP, create pads to send and receive data to and from the manager and to
2530 * intercept various events and queries
2533 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
2534 GstRTSPTransport * transport, GstPad ** outpad)
2537 GstPadTemplate *template;
2538 GstPad *pad0, *pad1;
2540 /* configure for interleaved delivery, nothing needs to be done
2541 * here, the loop function will call the chain functions of the
2542 * session manager. */
2543 stream->channel[0] = transport->interleaved.min;
2544 stream->channel[1] = transport->interleaved.max;
2545 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
2546 stream->channel[0], stream->channel[1]);
2548 /* we can remove the allocated UDP ports now */
2549 gst_rtspsrc_stream_free_udp (stream);
2551 /* no session manager, send data to srcpad directly */
2552 if (!stream->channelpad[0]) {
2553 GST_DEBUG_OBJECT (src, "no manager, creating pad");
2555 /* create a new pad we will use to stream to */
2556 name = g_strdup_printf ("stream_%u", stream->id);
2557 template = gst_static_pad_template_get (&rtptemplate);
2558 stream->channelpad[0] = gst_pad_new_from_template (template, name);
2559 gst_object_unref (template);
2562 /* set caps and activate */
2563 gst_pad_use_fixed_caps (stream->channelpad[0]);
2564 gst_pad_set_active (stream->channelpad[0], TRUE);
2566 *outpad = gst_object_ref (stream->channelpad[0]);
2568 GST_DEBUG_OBJECT (src, "using manager source pad");
2570 template = gst_static_pad_template_get (&anysrctemplate);
2572 /* allocate pads for sending the channel data into the manager */
2573 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
2574 gst_pad_link (pad0, stream->channelpad[0]);
2575 gst_object_unref (stream->channelpad[0]);
2576 stream->channelpad[0] = pad0;
2577 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
2578 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
2579 gst_pad_set_element_private (pad0, src);
2580 gst_pad_set_active (pad0, TRUE);
2582 if (stream->channelpad[1]) {
2583 /* if we have a sinkpad for the other channel, create a pad and link to the
2585 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
2586 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
2587 gst_pad_link (pad1, stream->channelpad[1]);
2588 gst_object_unref (stream->channelpad[1]);
2589 stream->channelpad[1] = pad1;
2590 gst_pad_set_active (pad1, TRUE);
2592 gst_object_unref (template);
2594 /* setup RTCP transport back to the server if we have to. */
2595 if (src->manager && src->do_rtcp) {
2598 template = gst_static_pad_template_get (&anysinktemplate);
2600 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
2601 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
2602 gst_pad_set_element_private (stream->rtcppad, stream);
2603 gst_pad_set_active (stream->rtcppad, TRUE);
2605 /* get session RTCP pad */
2606 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2607 pad = gst_element_get_request_pad (src->manager, name);
2612 gst_pad_link (pad, stream->rtcppad);
2613 gst_object_unref (pad);
2616 gst_object_unref (template);
2622 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
2623 GstRTSPTransport * transport, const gchar ** destination, gint * min,
2624 gint * max, guint * ttl)
2626 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
2628 if (!(*destination = transport->destination))
2629 *destination = stream->destination;
2632 /* transport first */
2633 *min = transport->port.min;
2634 *max = transport->port.max;
2635 if (*min == -1 && *max == -1) {
2636 /* then try from SDP */
2637 if (stream->port != 0) {
2638 *min = stream->port;
2639 *max = stream->port + 1;
2645 if (!(*ttl = transport->ttl))
2650 /* first take the source, then the endpoint to figure out where to send
2652 if (!(*destination = transport->source)) {
2653 if (src->conninfo.connection)
2654 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
2655 else if (stream->conninfo.connection)
2657 gst_rtsp_connection_get_ip (stream->conninfo.connection);
2661 /* for unicast we only expect the ports here */
2662 *min = transport->server_port.min;
2663 *max = transport->server_port.max;
2668 /* For multicast create UDP sources and join the multicast group. */
2670 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
2671 GstRTSPTransport * transport, GstPad ** outpad)
2674 const gchar *destination;
2677 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
2679 /* we can remove the allocated UDP ports now */
2680 gst_rtspsrc_stream_free_udp (stream);
2682 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
2685 /* we need a destination now */
2686 if (destination == NULL)
2687 goto no_destination;
2689 /* we really need ports now or we won't be able to receive anything at all */
2690 if (min == -1 && max == -1)
2693 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
2694 destination, min, max);
2696 /* creating UDP source for RTP */
2698 uri = g_strdup_printf ("udp://%s:%d", destination, min);
2699 stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
2701 if (stream->udpsrc[0] == NULL)
2704 /* take ownership */
2705 gst_object_ref_sink (stream->udpsrc[0]);
2708 gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
2711 /* creating another UDP source for RTCP */
2713 uri = g_strdup_printf ("udp://%s:%d", destination, max);
2714 stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
2716 if (stream->udpsrc[1] == NULL)
2719 /* take ownership */
2720 gst_object_ref_sink (stream->udpsrc[1]);
2722 gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
2729 GST_DEBUG_OBJECT (src, "no UDP source element found");
2734 GST_DEBUG_OBJECT (src, "no destination found");
2739 GST_DEBUG_OBJECT (src, "no ports found");
2744 /* configure the remainder of the UDP ports */
2746 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
2747 GstRTSPTransport * transport, GstPad ** outpad)
2749 /* we manage the UDP elements now. For unicast, the UDP sources where
2750 * allocated in the stream when we suggested a transport. */
2751 if (stream->udpsrc[0]) {
2752 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
2754 GST_DEBUG_OBJECT (src, "setting up UDP source");
2756 /* configure a timeout on the UDP port. When the timeout message is
2757 * posted, we assume UDP transport is not possible. We reconnect using TCP
2759 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->udp_timeout,
2762 /* get output pad of the UDP source. */
2763 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
2765 /* save it so we can unblock */
2766 stream->blockedpad = *outpad;
2768 /* configure pad block on the pad. As soon as there is dataflow on the
2769 * UDP source, we know that UDP is not blocked by a firewall and we can
2770 * configure all the streams to let the application autoplug decoders. */
2772 gst_pad_add_probe (stream->blockedpad,
2773 GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM, pad_blocked, src, NULL);
2775 if (stream->channelpad[0]) {
2776 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
2777 /* configure for UDP delivery, we need to connect the UDP pads to
2778 * the session plugin. */
2779 gst_pad_link (*outpad, stream->channelpad[0]);
2780 gst_object_unref (*outpad);
2782 /* we connected to pad-added signal to get pads from the manager */
2784 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
2789 if (stream->udpsrc[1]) {
2790 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
2792 if (stream->channelpad[1]) {
2795 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
2797 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
2798 gst_pad_link (pad, stream->channelpad[1]);
2799 gst_object_unref (pad);
2801 /* leave unlinked */
2807 /* configure the UDP sink back to the server for status reports */
2809 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
2810 GstRTSPStream * stream, GstRTSPTransport * transport)
2813 gint rtp_port, rtcp_port, sockfd = -1;
2814 gboolean do_rtp, do_rtcp;
2815 const gchar *destination;
2819 /* get transport info */
2820 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
2821 &rtp_port, &rtcp_port, &ttl);
2823 /* see what we need to do */
2824 do_rtp = (rtp_port != -1);
2825 /* it's possible that the server does not want us to send RTCP in which case
2827 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
2829 /* we need a destination when we have RTP or RTCP ports */
2830 if (destination == NULL && (do_rtp || do_rtcp))
2831 goto no_destination;
2833 /* try to construct the fakesrc to the RTP port of the server to open up any
2836 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
2839 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
2840 stream->udpsink[0] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
2842 if (stream->udpsink[0] == NULL)
2843 goto no_sink_element;
2845 /* don't join multicast group, we will have the source socket do that */
2846 /* no sync or async state changes needed */
2847 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
2848 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2850 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2852 if (stream->udpsrc[0]) {
2853 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
2854 * so that NAT firewalls will open a hole for us */
2855 g_object_get (G_OBJECT (stream->udpsrc[0]), "sock", &sockfd, NULL);
2856 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %d", sockfd);
2857 /* configure socket and make sure udpsink does not close it when shutting
2858 * down, it belongs to udpsrc after all. */
2859 g_object_set (G_OBJECT (stream->udpsink[0]), "sockfd", sockfd,
2860 "closefd", FALSE, NULL);
2863 /* the source for the dummy packets to open up NAT */
2864 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
2865 if (stream->fakesrc == NULL)
2866 goto no_fakesrc_element;
2868 /* random data in 5 buffers, a size of 200 bytes should be fine */
2869 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
2870 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
2872 /* we don't want to consider this a sink */
2873 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
2875 /* keep everything locked */
2876 gst_element_set_locked_state (stream->udpsink[0], TRUE);
2877 gst_element_set_locked_state (stream->fakesrc, TRUE);
2879 gst_object_ref (stream->udpsink[0]);
2880 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
2881 gst_object_ref (stream->fakesrc);
2882 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
2884 gst_element_link (stream->fakesrc, stream->udpsink[0]);
2887 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
2890 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
2891 stream->udpsink[1] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
2893 if (stream->udpsink[1] == NULL)
2894 goto no_sink_element;
2896 /* don't join multicast group, we will have the source socket do that */
2897 /* no sync or async state changes needed */
2898 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
2899 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
2901 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
2903 if (stream->udpsrc[1]) {
2904 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
2905 * because some servers check the port number of where it sends RTCP to identify
2906 * the RTCP packets it receives */
2907 g_object_get (G_OBJECT (stream->udpsrc[1]), "sock", &sockfd, NULL);
2908 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %d", sockfd);
2909 /* configure socket and make sure udpsink does not close it when shutting
2910 * down, it belongs to udpsrc after all. */
2911 g_object_set (G_OBJECT (stream->udpsink[1]), "sockfd", sockfd,
2912 "closefd", FALSE, NULL);
2915 /* we don't want to consider this a sink */
2916 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
2918 /* we keep this playing always */
2919 gst_element_set_locked_state (stream->udpsink[1], TRUE);
2920 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
2922 gst_object_ref (stream->udpsink[1]);
2923 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
2925 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
2927 /* get session RTCP pad */
2928 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
2929 pad = gst_element_get_request_pad (src->manager, name);
2934 gst_pad_link (pad, stream->rtcppad);
2935 gst_object_unref (pad);
2944 GST_DEBUG_OBJECT (src, "no destination address specified");
2949 GST_DEBUG_OBJECT (src, "no UDP sink element found");
2954 GST_DEBUG_OBJECT (src, "no fakesrc element found");
2959 /* sets up all elements needed for streaming over the specified transport.
2960 * Does not yet expose the element pads, this will be done when there is actuall
2961 * dataflow detected, which might never happen when UDP is blocked in a
2962 * firewall, for example.
2965 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
2966 GstRTSPTransport * transport)
2969 GstPad *outpad = NULL;
2970 GstPadTemplate *template;
2975 src = stream->parent;
2977 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
2979 s = gst_caps_get_structure (stream->caps, 0);
2981 /* get the proper mime type for this stream now */
2982 if (gst_rtsp_transport_get_mime (transport->trans, &mime) < 0)
2983 goto unknown_transport;
2985 goto unknown_transport;
2987 /* configure the final mime type */
2988 GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
2989 gst_structure_set_name (s, mime);
2991 /* try to get and configure a manager, channelpad[0-1] will be configured with
2992 * the pads for the manager, or NULL when no manager is needed. */
2993 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
2996 switch (transport->lower_transport) {
2997 case GST_RTSP_LOWER_TRANS_TCP:
2998 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
2999 goto transport_failed;
3001 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3002 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
3003 goto transport_failed;
3004 /* fallthrough, the rest is the same for UDP and MCAST */
3005 case GST_RTSP_LOWER_TRANS_UDP:
3006 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
3007 goto transport_failed;
3008 /* configure udpsinks back to the server for RTCP messages and for the
3009 * dummy RTP messages to open NAT. */
3010 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
3011 goto transport_failed;
3014 goto unknown_transport;
3018 GST_DEBUG_OBJECT (src, "creating ghostpad");
3020 gst_pad_use_fixed_caps (outpad);
3022 /* create ghostpad, don't add just yet, this will be done when we activate
3024 name = g_strdup_printf ("stream_%u", stream->id);
3025 template = gst_static_pad_template_get (&rtptemplate);
3026 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
3027 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
3028 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
3029 gst_object_unref (template);
3032 gst_object_unref (outpad);
3034 /* mark pad as ok */
3035 stream->last_ret = GST_FLOW_OK;
3042 GST_DEBUG_OBJECT (src, "failed to configure transport");
3047 GST_DEBUG_OBJECT (src, "unknown transport");
3052 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3057 /* send a couple of dummy random packets on the receiver RTP port to the server,
3058 * this should make a firewall think we initiated the data transfer and
3059 * hopefully allow packets to go from the sender port to our RTP receiver port */
3061 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
3065 if (src->nat_method != GST_RTSP_NAT_DUMMY)
3068 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3069 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3071 if (stream->fakesrc && stream->udpsink[0]) {
3072 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
3073 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
3074 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
3075 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
3076 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
3082 /* Adds the source pads of all configured streams to the element.
3083 * This code is performed when we detected dataflow.
3085 * We detect dataflow from either the _loop function or with pad probes on the
3089 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
3093 GST_DEBUG_OBJECT (src, "activating streams");
3095 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3096 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3098 if (stream->udpsrc[0]) {
3099 /* remove timeout, we are streaming now and timeouts will be handled by
3100 * the session manager and jitter buffer */
3101 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
3103 if (stream->srcpad) {
3104 /* if we don't have a session manager, set the caps now. If we have a
3105 * session, we will get a notification of the pad and the caps. */
3106 if (!src->manager) {
3107 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
3108 gst_pad_set_caps (stream->srcpad, stream->caps);
3111 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
3112 gst_pad_set_active (stream->srcpad, TRUE);
3114 if (!stream->added) {
3115 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
3116 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
3117 stream->added = TRUE;
3122 /* unblock all pads */
3123 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3124 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3126 if (stream->blockid) {
3127 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
3128 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
3129 stream->blockid = 0;
3137 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment)
3140 guint64 start, stop;
3141 gdouble play_speed, play_scale;
3143 GST_DEBUG_OBJECT (src, "configuring stream caps");
3145 start = segment->position;
3146 stop = segment->duration;
3147 play_speed = segment->rate;
3148 play_scale = segment->applied_rate;
3150 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3151 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3154 if ((caps = stream->caps)) {
3155 caps = gst_caps_make_writable (caps);
3157 if (stream->timebase != -1)
3158 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
3159 (guint) stream->timebase, NULL);
3160 if (stream->seqbase != -1)
3161 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
3162 (guint) stream->seqbase, NULL);
3163 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
3165 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
3166 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
3167 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
3169 stream->caps = caps;
3171 GST_DEBUG_OBJECT (src, "stream %p, caps %" GST_PTR_FORMAT, stream, caps);
3174 GST_DEBUG_OBJECT (src, "clear session");
3175 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
3179 static GstFlowReturn
3180 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
3185 /* store the value */
3186 stream->last_ret = ret;
3188 /* if it's success we can return the value right away */
3189 if (ret == GST_FLOW_OK)
3192 /* any other error that is not-linked can be returned right
3194 if (ret != GST_FLOW_NOT_LINKED)
3197 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
3198 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3199 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3201 ret = ostream->last_ret;
3202 /* some other return value (must be SUCCESS but we can return
3203 * other values as well) */
3204 if (ret != GST_FLOW_NOT_LINKED)
3207 /* if we get here, all other pads were unlinked and we return
3208 * NOT_LINKED then */
3214 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
3215 GstEvent * event, gboolean source)
3217 gboolean res = TRUE;
3219 /* only streams that have a connection to the outside world */
3220 if (stream->srcpad == NULL)
3223 if (source && stream->udpsrc[0]) {
3224 gst_event_ref (event);
3225 res = gst_element_send_event (stream->udpsrc[0], event);
3226 } else if (stream->channelpad[0]) {
3227 gst_event_ref (event);
3228 if (GST_PAD_IS_SRC (stream->channelpad[0]))
3229 res = gst_pad_push_event (stream->channelpad[0], event);
3231 res = gst_pad_send_event (stream->channelpad[0], event);
3234 if (source && stream->udpsrc[1]) {
3235 gst_event_ref (event);
3236 res &= gst_element_send_event (stream->udpsrc[1], event);
3237 } else if (stream->channelpad[1]) {
3238 gst_event_ref (event);
3239 if (GST_PAD_IS_SRC (stream->channelpad[1]))
3240 res &= gst_pad_push_event (stream->channelpad[1], event);
3242 res &= gst_pad_send_event (stream->channelpad[1], event);
3246 gst_event_unref (event);
3252 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event, gboolean source)
3255 gboolean res = TRUE;
3257 for (streams = src->streams; streams; streams = g_list_next (streams)) {
3258 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
3260 gst_event_ref (event);
3261 res &= gst_rtspsrc_stream_push_event (src, ostream, event, source);
3263 gst_event_unref (event);
3268 static GstRTSPResult
3269 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3274 if (info->connection == NULL) {
3275 if (info->url == NULL) {
3276 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
3277 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
3281 /* create connection */
3282 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
3283 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
3284 goto could_not_create;
3287 g_free (info->url_str);
3288 info->url_str = gst_rtsp_url_get_request_uri (info->url);
3290 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
3292 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
3293 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
3295 if (src->proxy_host) {
3296 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
3298 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
3303 if (!info->connected) {
3306 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
3307 ("Connecting to %s", info->location));
3308 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
3310 gst_rtsp_connection_connect (info->connection,
3311 src->ptcp_timeout)) < 0)
3312 goto could_not_connect;
3314 info->connected = TRUE;
3321 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
3326 gchar *str = gst_rtsp_strresult (res);
3327 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
3333 gchar *str = gst_rtsp_strresult (res);
3334 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
3340 static GstRTSPResult
3341 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
3344 if (info->connected) {
3345 GST_DEBUG_OBJECT (src, "closing connection...");
3346 gst_rtsp_connection_close (info->connection);
3347 info->connected = FALSE;
3349 if (free && info->connection) {
3350 /* free connection */
3351 GST_DEBUG_OBJECT (src, "freeing connection...");
3352 gst_rtsp_connection_free (info->connection);
3353 info->connection = NULL;
3358 static GstRTSPResult
3359 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
3364 GST_DEBUG_OBJECT (src, "reconnecting connection...");
3365 gst_rtsp_conninfo_close (src, info, FALSE);
3366 res = gst_rtsp_conninfo_connect (src, info, async);
3372 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
3376 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
3377 if (src->conninfo.connection) {
3378 GST_DEBUG_OBJECT (src, "connection flush");
3379 gst_rtsp_connection_flush (src->conninfo.connection, flush);
3381 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3382 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3383 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
3384 if (stream->conninfo.connection)
3385 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
3389 /* FIXME, handle server request, reply with OK, for now */
3390 static GstRTSPResult
3391 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
3392 GstRTSPMessage * request)
3394 GstRTSPMessage response = { 0 };
3397 GST_DEBUG_OBJECT (src, "got server request message");
3400 gst_rtsp_message_dump (request);
3402 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
3404 if (res == GST_RTSP_ENOTIMPL) {
3405 /* default implementation, send OK */
3407 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
3412 GST_DEBUG_OBJECT (src, "replying with OK");
3415 gst_rtsp_message_dump (&response);
3417 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
3421 gst_rtsp_message_unset (&response);
3422 } else if (res == GST_RTSP_EEOF)
3430 gst_rtsp_message_unset (&response);
3435 /* send server keep-alive */
3436 static GstRTSPResult
3437 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
3439 GstRTSPMessage request = { 0 };
3441 GstRTSPMethod method;
3444 GST_DEBUG_OBJECT (src, "creating server keep-alive");
3446 /* find a method to use for keep-alive */
3447 if (src->methods & GST_RTSP_GET_PARAMETER)
3448 method = GST_RTSP_GET_PARAMETER;
3450 method = GST_RTSP_OPTIONS;
3453 control = src->control;
3455 control = src->conninfo.url_str;
3457 if (control == NULL)
3460 res = gst_rtsp_message_init_request (&request, method, control);
3465 gst_rtsp_message_dump (&request);
3468 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
3473 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
3474 gst_rtsp_message_unset (&request);
3481 GST_WARNING_OBJECT (src, "no control url to send keepalive");
3486 gchar *str = gst_rtsp_strresult (res);
3488 gst_rtsp_message_unset (&request);
3489 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
3490 ("Could not send keep-alive. (%s)", str));
3496 static GstFlowReturn
3497 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
3499 GstRTSPMessage message = { 0 };
3502 GstRTSPStream *stream;
3503 GstPad *outpad = NULL;
3506 GstFlowReturn ret = GST_FLOW_OK;
3508 gboolean is_rtcp, have_data;
3510 /* here we are only interested in data messages */
3513 GTimeVal tv_timeout;
3515 /* get the next timeout interval */
3516 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3518 /* see if the timeout period expired */
3519 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
3520 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
3521 /* send keep-alive, only act on interrupt, a warning will be posted for
3523 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3525 /* get new timeout */
3526 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3529 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
3530 tv_timeout.tv_sec, tv_timeout.tv_usec);
3532 /* protect the connection with the connection lock so that we can see when
3533 * we are finished doing server communication */
3535 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3536 &message, src->ptcp_timeout);
3540 GST_DEBUG_OBJECT (src, "we received a server message");
3542 case GST_RTSP_EINTR:
3543 /* we got interrupted this means we need to stop */
3545 case GST_RTSP_ETIMEOUT:
3546 /* no reply, send keep alive */
3547 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3548 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3552 /* go EOS when the server closed the connection */
3558 switch (message.type) {
3559 case GST_RTSP_MESSAGE_REQUEST:
3560 /* server sends us a request message, handle it */
3562 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3564 if (res == GST_RTSP_EEOF)
3567 goto handle_request_failed;
3569 case GST_RTSP_MESSAGE_RESPONSE:
3570 /* we ignore response messages */
3571 GST_DEBUG_OBJECT (src, "ignoring response message");
3573 gst_rtsp_message_dump (&message);
3575 case GST_RTSP_MESSAGE_DATA:
3576 GST_DEBUG_OBJECT (src, "got data message");
3580 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3587 channel = message.type_data.data.channel;
3589 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
3591 goto unknown_stream;
3593 if (channel == stream->channel[0]) {
3594 outpad = stream->channelpad[0];
3596 } else if (channel == stream->channel[1]) {
3597 outpad = stream->channelpad[1];
3603 /* take a look at the body to figure out what we have */
3604 gst_rtsp_message_get_body (&message, &data, &size);
3606 goto invalid_length;
3608 /* channels are not correct on some servers, do extra check */
3609 if (data[1] >= 200 && data[1] <= 204) {
3610 /* hmm RTCP message switch to the RTCP pad of the same stream. */
3611 outpad = stream->channelpad[1];
3615 /* we have no clue what this is, just ignore then. */
3617 goto unknown_stream;
3619 /* take the message body for further processing */
3620 gst_rtsp_message_steal_body (&message, &data, &size);
3622 /* strip the trailing \0 */
3625 buf = gst_buffer_new ();
3626 gst_buffer_take_memory (buf, -1,
3627 gst_memory_new_wrapped (0, data, g_free, size, 0, size));
3629 /* don't need message anymore */
3630 gst_rtsp_message_unset (&message);
3632 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
3635 if (src->need_activate) {
3636 gst_rtspsrc_activate_streams (src);
3637 src->need_activate = FALSE;
3640 if (src->base_time == -1) {
3641 /* Take current running_time. This timestamp will be put on
3642 * the first buffer of each stream because we are a live source and so we
3643 * timestamp with the running_time. When we are dealing with TCP, we also
3644 * only timestamp the first buffer (using the DISCONT flag) because a server
3645 * typically bursts data, for which we don't want to compensate by speeding
3646 * up the media. The other timestamps will be interpollated from this one
3647 * using the RTP timestamps. */
3648 GST_OBJECT_LOCK (src);
3649 if (GST_ELEMENT_CLOCK (src)) {
3651 GstClockTime base_time;
3653 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
3654 base_time = GST_ELEMENT_CAST (src)->base_time;
3656 src->base_time = now - base_time;
3658 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
3659 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
3661 GST_OBJECT_UNLOCK (src);
3664 if (stream->discont && !is_rtcp) {
3665 /* mark first RTP buffer as discont */
3666 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
3667 stream->discont = FALSE;
3668 /* first buffer gets the timestamp, other buffers are not timestamped and
3669 * their presentation time will be interpollated from the rtp timestamps. */
3670 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
3671 GST_TIME_ARGS (src->base_time));
3673 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
3676 /* chain to the peer pad */
3677 if (GST_PAD_IS_SINK (outpad))
3678 ret = gst_pad_chain (outpad, buf);
3680 ret = gst_pad_push (outpad, buf);
3683 /* combine all stream flows for the data transport */
3684 ret = gst_rtspsrc_combine_flows (src, stream, ret);
3691 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
3692 gst_rtsp_message_unset (&message);
3697 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3698 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3699 ("The server closed the connection."));
3700 src->conninfo.connected = FALSE;
3701 gst_rtsp_message_unset (&message);
3702 return GST_FLOW_UNEXPECTED;
3706 gst_rtsp_message_unset (&message);
3707 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3708 gst_rtspsrc_connection_flush (src, FALSE);
3709 return GST_FLOW_WRONG_STATE;
3713 gchar *str = gst_rtsp_strresult (res);
3715 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3716 ("Could not receive message. (%s)", str));
3719 gst_rtsp_message_unset (&message);
3720 return GST_FLOW_ERROR;
3722 handle_request_failed:
3724 gchar *str = gst_rtsp_strresult (res);
3726 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3727 ("Could not handle server message. (%s)", str));
3729 gst_rtsp_message_unset (&message);
3730 return GST_FLOW_ERROR;
3734 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3735 ("Short message received, ignoring."));
3736 gst_rtsp_message_unset (&message);
3741 static GstFlowReturn
3742 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
3745 GstRTSPMessage message = { 0 };
3749 GTimeVal tv_timeout;
3751 /* get the next timeout interval */
3752 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
3754 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
3755 (gint) tv_timeout.tv_sec);
3757 gst_rtsp_message_unset (&message);
3759 /* we should continue reading the TCP socket because the server might
3760 * send us requests. When the session timeout expires, we need to send a
3761 * keep-alive request to keep the session open. */
3762 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
3763 &message, &tv_timeout);
3767 GST_DEBUG_OBJECT (src, "we received a server message");
3769 case GST_RTSP_EINTR:
3770 /* we got interrupted, see what we have to do */
3772 case GST_RTSP_ETIMEOUT:
3773 /* send keep-alive, ignore the result, a warning will be posted. */
3774 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
3775 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3779 /* server closed the connection. not very fatal for UDP, reconnect and
3780 * see what happens. */
3781 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3782 ("The server closed the connection."));
3784 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
3792 switch (message.type) {
3793 case GST_RTSP_MESSAGE_REQUEST:
3794 /* server sends us a request message, handle it */
3796 gst_rtspsrc_handle_request (src, src->conninfo.connection,
3798 if (res == GST_RTSP_EEOF)
3801 goto handle_request_failed;
3803 case GST_RTSP_MESSAGE_RESPONSE:
3804 /* we ignore response and data messages */
3805 GST_DEBUG_OBJECT (src, "ignoring response message");
3807 gst_rtsp_message_dump (&message);
3808 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
3809 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
3810 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
3811 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
3812 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
3819 case GST_RTSP_MESSAGE_DATA:
3820 /* we ignore response and data messages */
3821 GST_DEBUG_OBJECT (src, "ignoring data message");
3824 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
3830 /* we get here when the connection got interrupted */
3833 gst_rtsp_message_unset (&message);
3834 GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
3835 gst_rtspsrc_connection_flush (src, FALSE);
3836 return GST_FLOW_WRONG_STATE;
3840 gchar *str = gst_rtsp_strresult (res);
3843 src->conninfo.connected = FALSE;
3844 if (res != GST_RTSP_EINTR) {
3845 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
3846 ("Could not connect to server. (%s)", str));
3848 ret = GST_FLOW_ERROR;
3850 ret = GST_FLOW_WRONG_STATE;
3856 gchar *str = gst_rtsp_strresult (res);
3858 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3859 ("Could not receive message. (%s)", str));
3861 return GST_FLOW_ERROR;
3863 handle_request_failed:
3865 gchar *str = gst_rtsp_strresult (res);
3868 gst_rtsp_message_unset (&message);
3869 if (res != GST_RTSP_EINTR) {
3870 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
3871 ("Could not handle server message. (%s)", str));
3873 ret = GST_FLOW_ERROR;
3875 ret = GST_FLOW_WRONG_STATE;
3881 GST_DEBUG_OBJECT (src, "we got an eof from the server");
3882 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3883 ("The server closed the connection."));
3884 src->conninfo.connected = FALSE;
3885 gst_rtsp_message_unset (&message);
3886 return GST_FLOW_UNEXPECTED;
3890 static GstRTSPResult
3891 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
3893 GstRTSPResult res = GST_RTSP_OK;
3896 GST_DEBUG_OBJECT (src, "doing reconnect");
3898 GST_OBJECT_LOCK (src);
3899 /* only restart when the pads were not yet activated, else we were
3900 * streaming over UDP */
3901 restart = src->need_activate;
3902 GST_OBJECT_UNLOCK (src);
3904 /* no need to restart, we're done */
3908 /* we can try only TCP now */
3909 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
3911 /* close and cleanup our state */
3912 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
3915 /* see if we have TCP left to try. Also don't try TCP when we were configured
3917 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
3920 /* We post a warning message now to inform the user
3921 * that nothing happened. It's most likely a firewall thing. */
3922 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
3923 ("Could not receive any UDP packets for %.4f seconds, maybe your "
3924 "firewall is blocking it. Retrying using a TCP connection.",
3925 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
3927 /* open new connection using tcp */
3928 if (gst_rtspsrc_open (src, async) < 0)
3931 /* start playback */
3932 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
3941 src->cur_protocols = 0;
3942 /* no transport possible, post an error and stop */
3943 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
3944 ("Could not receive any UDP packets for %.4f seconds, maybe your "
3945 "firewall is blocking it. No other protocols to try.",
3946 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
3947 return GST_FLOW_ERROR;
3951 GST_DEBUG_OBJECT (src, "open failed");
3956 GST_DEBUG_OBJECT (src, "play failed");
3962 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
3966 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
3969 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
3972 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
3975 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
3983 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
3987 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
3990 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
3993 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
3996 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
4004 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
4008 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
4011 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
4014 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
4017 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
4025 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
4029 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
4032 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
4035 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
4038 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
4046 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
4048 if (ret == GST_RTSP_OK)
4049 gst_rtspsrc_loop_complete_cmd (src, cmd);
4050 else if (ret == GST_RTSP_EINTR)
4051 gst_rtspsrc_loop_cancel_cmd (src, cmd);
4053 gst_rtspsrc_loop_error_cmd (src, cmd);
4057 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush)
4061 /* FIXME flush param mute; remove at discretion */
4063 /* start new request */
4064 gst_rtspsrc_loop_start_cmd (src, cmd);
4066 GST_DEBUG_OBJECT (src, "sending cmd %d", cmd);
4068 GST_OBJECT_LOCK (src);
4069 old = src->loop_cmd;
4070 if (old != CMD_WAIT) {
4071 src->loop_cmd = CMD_WAIT;
4072 GST_OBJECT_UNLOCK (src);
4073 /* cancel previous request */
4074 gst_rtspsrc_loop_cancel_cmd (src, old);
4075 GST_OBJECT_LOCK (src);
4077 src->loop_cmd = cmd;
4078 /* interrupt if allowed */
4080 GST_DEBUG_OBJECT (src, "start connection flush");
4081 gst_rtspsrc_connection_flush (src, TRUE);
4084 gst_task_start (src->task);
4085 GST_OBJECT_UNLOCK (src);
4089 gst_rtspsrc_loop (GstRTSPSrc * src)
4093 if (!src->conninfo.connection || !src->conninfo.connected)
4096 if (src->interleaved)
4097 ret = gst_rtspsrc_loop_interleaved (src);
4099 ret = gst_rtspsrc_loop_udp (src);
4101 if (ret != GST_FLOW_OK)
4109 GST_WARNING_OBJECT (src, "we are not connected");
4110 ret = GST_FLOW_WRONG_STATE;
4115 const gchar *reason = gst_flow_get_name (ret);
4117 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
4118 src->running = FALSE;
4119 if (ret == GST_FLOW_UNEXPECTED) {
4120 /* perform EOS logic */
4121 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
4122 gst_element_post_message (GST_ELEMENT_CAST (src),
4123 gst_message_new_segment_done (GST_OBJECT_CAST (src),
4124 src->segment.format, src->segment.position));
4126 gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
4128 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_UNEXPECTED) {
4129 /* for fatal errors we post an error message, post the error before the
4130 * EOS so the app knows about the error first. */
4131 GST_ELEMENT_ERROR (src, STREAM, FAILED,
4132 ("Internal data flow error."),
4133 ("streaming task paused, reason %s (%d)", reason, ret));
4134 gst_rtspsrc_push_event (src, gst_event_new_eos (), FALSE);
4140 #ifndef GST_DISABLE_GST_DEBUG
4141 static const gchar *
4142 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
4146 while (method != 0) {
4163 static const gchar *
4164 gst_rtspsrc_skip_lws (const gchar * s)
4166 while (g_ascii_isspace (*s))
4171 static const gchar *
4172 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
4174 while (s > start && g_ascii_isspace (*(s - 1)))
4179 static const gchar *
4180 gst_rtspsrc_skip_commas (const gchar * s)
4182 /* The grammar allows for multiple commas */
4183 while (g_ascii_isspace (*s) || *s == ',')
4188 static const gchar *
4189 gst_rtspsrc_skip_item (const gchar * s)
4191 gboolean quoted = FALSE;
4192 const gchar *start = s;
4194 /* A list item ends at the last non-whitespace character
4195 * before a comma which is not inside a quoted-string. Or at
4196 * the end of the string.
4202 if (*s == '\\' && *(s + 1))
4211 return gst_rtspsrc_unskip_lws (s, start);
4215 gst_rtsp_decode_quoted_string (gchar * quoted_string)
4219 src = quoted_string + 1;
4220 dst = quoted_string;
4221 while (*src && *src != '"') {
4222 if (*src == '\\' && *(src + 1))
4229 /* Extract the authentication tokens that the server provided for each method
4230 * into an array of structures and give those to the connection object.
4233 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
4234 const gchar * header, gboolean * stale)
4236 GSList *list = NULL, *iter;
4238 gchar *item, *eq, *name_end, *value;
4240 g_return_if_fail (stale != NULL);
4242 gst_rtsp_connection_clear_auth_params (conn);
4245 /* Parse a header whose content is described by RFC2616 as
4246 * "#something", where "something" does not itself contain commas,
4247 * except as part of quoted-strings, into a list of allocated strings.
4249 header = gst_rtspsrc_skip_commas (header);
4251 end = gst_rtspsrc_skip_item (header);
4252 list = g_slist_prepend (list, g_strndup (header, end - header));
4253 header = gst_rtspsrc_skip_commas (end);
4258 list = g_slist_reverse (list);
4259 for (iter = list; iter; iter = iter->next) {
4262 eq = strchr (item, '=');
4264 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
4265 if (name_end == item) {
4266 /* That's no good... */
4273 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
4275 gst_rtsp_decode_quoted_string (value);
4279 if ((strcmp (item, "stale") == 0) && (strcmp (value, "TRUE") == 0))
4281 gst_rtsp_connection_set_auth_param (conn, item, value);
4285 g_slist_free (list);
4288 /* Parse a WWW-Authenticate Response header and determine the
4289 * available authentication methods
4291 * This code should also cope with the fact that each WWW-Authenticate
4292 * header can contain multiple challenge methods + tokens
4294 * At the moment, for Basic auth, we just do a minimal check and don't
4295 * even parse out the realm */
4297 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
4298 GstRTSPConnection * conn, gboolean * stale)
4302 g_return_if_fail (hdr != NULL);
4303 g_return_if_fail (methods != NULL);
4304 g_return_if_fail (stale != NULL);
4306 /* Skip whitespace at the start of the string */
4307 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
4309 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
4310 *methods |= GST_RTSP_AUTH_BASIC;
4311 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
4312 *methods |= GST_RTSP_AUTH_DIGEST;
4313 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
4318 * gst_rtspsrc_setup_auth:
4319 * @src: the rtsp source
4321 * Configure a username and password and auth method on the
4322 * connection object based on a response we received from the
4325 * Currently, this requires that a username and password were supplied
4326 * in the uri. In the future, they may be requested on demand by sending
4327 * a message up the bus.
4329 * Returns: TRUE if authentication information could be set up correctly.
4332 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
4336 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
4337 GstRTSPAuthMethod method;
4338 GstRTSPResult auth_result;
4340 GstRTSPConnection *conn;
4342 gboolean stale = FALSE;
4344 conn = src->conninfo.connection;
4346 /* Identify the available auth methods and see if any are supported */
4347 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
4348 &hdr, 0) == GST_RTSP_OK) {
4349 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
4352 if (avail_methods == GST_RTSP_AUTH_NONE)
4353 goto no_auth_available;
4355 /* For digest auth, if the response indicates that the session
4356 * data are stale, we just update them in the connection object and
4357 * return TRUE to retry the request */
4359 src->tried_url_auth = FALSE;
4361 url = gst_rtsp_connection_get_url (conn);
4363 /* Do we have username and password available? */
4364 if (url != NULL && !src->tried_url_auth && url->user != NULL
4365 && url->passwd != NULL) {
4368 src->tried_url_auth = TRUE;
4369 GST_DEBUG_OBJECT (src,
4370 "Attempting authentication using credentials from the URL");
4372 user = src->user_id;
4373 pass = src->user_pw;
4374 GST_DEBUG_OBJECT (src,
4375 "Attempting authentication using credentials from the properties");
4378 /* FIXME: If the url didn't contain username and password or we tried them
4379 * already, request a username and passwd from the application via some kind
4380 * of credentials request message */
4382 /* If we don't have a username and passwd at this point, bail out. */
4383 if (user == NULL || pass == NULL)
4386 /* Try to configure for each available authentication method, strongest to
4388 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
4389 /* Check if this method is available on the server */
4390 if ((method & avail_methods) == 0)
4393 /* Pass the credentials to the connection to try on the next request */
4394 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
4395 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
4396 * ignore it and end up retrying later */
4397 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
4398 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
4399 gst_rtsp_auth_method_to_string (method));
4404 if (method == GST_RTSP_AUTH_NONE)
4405 goto no_auth_available;
4411 /* Output an error indicating that we couldn't connect because there were
4412 * no supported authentication protocols */
4413 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4414 ("No supported authentication protocol was found"));
4419 /* We don't fire an error message, we just return FALSE and let the
4420 * normal NOT_AUTHORIZED error be propagated */
4425 static GstRTSPResult
4426 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4427 GstRTSPMessage * request, GstRTSPMessage * response,
4428 GstRTSPStatusCode * code)
4431 GstRTSPStatusCode thecode;
4432 gchar *content_base = NULL;
4436 if (!src->short_header)
4437 gst_rtsp_ext_list_before_send (src->extensions, request);
4439 GST_DEBUG_OBJECT (src, "sending message");
4442 gst_rtsp_message_dump (request);
4444 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
4448 gst_rtsp_connection_reset_timeout (conn);
4451 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
4456 gst_rtsp_message_dump (response);
4458 switch (response->type) {
4459 case GST_RTSP_MESSAGE_REQUEST:
4460 res = gst_rtspsrc_handle_request (src, conn, response);
4461 if (res == GST_RTSP_EEOF)
4464 goto handle_request_failed;
4466 case GST_RTSP_MESSAGE_RESPONSE:
4467 /* ok, a response is good */
4468 GST_DEBUG_OBJECT (src, "received response message");
4470 case GST_RTSP_MESSAGE_DATA:
4471 /* get next response */
4472 GST_DEBUG_OBJECT (src, "ignoring data response message");
4475 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4480 thecode = response->type_data.response.code;
4482 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
4484 /* if the caller wanted the result code, we store it. */
4488 /* If the request didn't succeed, bail out before doing any more */
4489 if (thecode != GST_RTSP_STS_OK)
4492 /* store new content base if any */
4493 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
4496 g_free (src->content_base);
4497 src->content_base = g_strdup (content_base);
4499 gst_rtsp_ext_list_after_send (src->extensions, request, response);
4506 gchar *str = gst_rtsp_strresult (res);
4508 if (res != GST_RTSP_EINTR) {
4509 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4510 ("Could not send message. (%s)", str));
4512 GST_WARNING_OBJECT (src, "send interrupted");
4521 GST_WARNING_OBJECT (src, "server closed connection, doing reconnect");
4524 /* if reconnect succeeds, try again */
4526 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
4530 /* only try once after reconnect, then fallthrough and error out */
4533 gchar *str = gst_rtsp_strresult (res);
4535 if (res != GST_RTSP_EINTR) {
4536 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4537 ("Could not receive message. (%s)", str));
4539 GST_WARNING_OBJECT (src, "receive interrupted");
4547 handle_request_failed:
4549 /* ERROR was posted */
4550 gst_rtsp_message_unset (response);
4555 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4556 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4557 ("The server closed the connection."));
4558 gst_rtsp_message_unset (response);
4565 * @src: the rtsp source
4566 * @conn: the connection to send on
4567 * @request: must point to a valid request
4568 * @response: must point to an empty #GstRTSPMessage
4569 * @code: an optional code result
4571 * send @request and retrieve the response in @response. optionally @code can be
4572 * non-NULL in which case it will contain the status code of the response.
4574 * If This function returns #GST_RTSP_OK, @response will contain a valid response
4575 * message that should be cleaned with gst_rtsp_message_unset() after usage.
4577 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
4578 * @response message) if the response code was not 200 (OK).
4580 * If the attempt results in an authentication failure, then this will attempt
4581 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
4584 * Returns: #GST_RTSP_OK if the processing was successful.
4586 static GstRTSPResult
4587 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
4588 GstRTSPMessage * request, GstRTSPMessage * response,
4589 GstRTSPStatusCode * code)
4591 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
4592 GstRTSPResult res = GST_RTSP_ERROR;
4595 GstRTSPMethod method = GST_RTSP_INVALID;
4601 /* make sure we don't loop forever */
4605 /* save method so we can disable it when the server complains */
4606 method = request->type_data.request.method;
4609 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
4613 case GST_RTSP_STS_UNAUTHORIZED:
4614 if (gst_rtspsrc_setup_auth (src, response)) {
4615 /* Try the request/response again after configuring the auth info
4623 } while (retry == TRUE);
4625 /* If the user requested the code, let them handle errors, otherwise
4626 * post an error below */
4629 else if (int_code != GST_RTSP_STS_OK)
4630 goto error_response;
4637 GST_DEBUG_OBJECT (src, "got error %d", res);
4642 res = GST_RTSP_ERROR;
4644 switch (response->type_data.response.code) {
4645 case GST_RTSP_STS_NOT_FOUND:
4646 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
4647 response->type_data.response.reason));
4649 case GST_RTSP_STS_MOVED_PERMANENTLY:
4650 case GST_RTSP_STS_MOVE_TEMPORARILY:
4652 gchar *new_location;
4653 GstRTSPLowerTrans transports;
4655 GST_DEBUG_OBJECT (src, "got redirection");
4656 /* if we don't have a Location Header, we must error */
4657 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
4658 &new_location, 0) < 0)
4661 /* When we receive a redirect result, we go back to the INIT state after
4662 * parsing the new URI. The caller should do the needed steps to issue
4663 * a new setup when it detects this state change. */
4664 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
4666 /* save current transports */
4667 if (src->conninfo.url)
4668 transports = src->conninfo.url->transports;
4670 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
4672 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
4674 /* set old transports */
4675 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
4676 src->conninfo.url->transports = transports;
4678 src->need_redirect = TRUE;
4679 src->state = GST_RTSP_STATE_INIT;
4683 case GST_RTSP_STS_NOT_ACCEPTABLE:
4684 case GST_RTSP_STS_NOT_IMPLEMENTED:
4685 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
4686 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
4687 gst_rtsp_method_as_text (method));
4688 src->methods &= ~method;
4692 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4693 ("Got error response: %d (%s).", response->type_data.response.code,
4694 response->type_data.response.reason));
4697 /* if we return ERROR we should unset the response ourselves */
4698 if (res == GST_RTSP_ERROR)
4699 gst_rtsp_message_unset (response);
4705 static GstRTSPResult
4706 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
4707 GstRTSPMessage * response, GstRTSPSrc * src)
4709 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
4714 /* parse the response and collect all the supported methods. We need this
4715 * information so that we don't try to send an unsupported request to the
4719 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
4721 GstRTSPHeaderField field;
4727 /* reset supported methods */
4730 /* Try Allow Header first */
4731 field = GST_RTSP_HDR_ALLOW;
4734 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4735 if (indx == 0 && !respoptions) {
4736 /* if no Allow header was found then try the Public header... */
4737 field = GST_RTSP_HDR_PUBLIC;
4738 gst_rtsp_message_get_header (response, field, &respoptions, indx);
4743 /* If we get here, the server gave a list of supported methods, parse
4744 * them here. The string is like:
4746 * OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
4748 options = g_strsplit (respoptions, ",", 0);
4750 for (i = 0; options[i]; i++) {
4754 stripped = g_strstrip (options[i]);
4755 method = gst_rtsp_find_method (stripped);
4757 /* keep bitfield of supported methods */
4758 if (method != GST_RTSP_INVALID)
4759 src->methods |= method;
4761 g_strfreev (options);
4766 if (src->methods == 0) {
4767 /* neither Allow nor Public are required, assume the server supports
4768 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
4770 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
4771 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
4773 /* always assume PLAY, FIXME, extensions should be able to override
4775 src->methods |= GST_RTSP_PLAY;
4776 /* also assume it will support Range */
4777 src->seekable = TRUE;
4779 /* we need describe and setup */
4780 if (!(src->methods & GST_RTSP_DESCRIBE))
4782 if (!(src->methods & GST_RTSP_SETUP))
4790 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4791 ("Server does not support DESCRIBE."));
4796 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
4797 ("Server does not support SETUP."));
4802 /* masks to be kept in sync with the hardcoded protocol order of preference
4804 static guint protocol_masks[] = {
4805 GST_RTSP_LOWER_TRANS_UDP,
4806 GST_RTSP_LOWER_TRANS_UDP_MCAST,
4807 GST_RTSP_LOWER_TRANS_TCP,
4811 static GstRTSPResult
4812 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
4813 GstRTSPLowerTrans protocols, gchar ** transports)
4817 gboolean add_udp_str;
4822 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
4827 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
4829 /* extension listed transports, use those */
4830 if (*transports != NULL)
4833 /* it's the default */
4834 add_udp_str = FALSE;
4836 /* the default RTSP transports */
4837 result = g_string_new ("");
4838 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
4839 GST_DEBUG_OBJECT (src, "adding UDP unicast");
4841 g_string_append (result, "RTP/AVP");
4843 g_string_append (result, "/UDP");
4844 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
4845 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
4846 GST_DEBUG_OBJECT (src, "adding UDP multicast");
4848 /* we don't have to allocate any UDP ports yet, if the selected transport
4849 * turns out to be multicast we can create them and join the multicast
4850 * group indicated in the transport reply */
4851 if (result->len > 0)
4852 g_string_append (result, ",");
4853 g_string_append (result, "RTP/AVP");
4855 g_string_append (result, "/UDP");
4856 g_string_append (result, ";multicast");
4857 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
4858 GST_DEBUG_OBJECT (src, "adding TCP");
4860 if (result->len > 0)
4861 g_string_append (result, ",");
4862 g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2");
4864 *transports = g_string_free (result, FALSE);
4866 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
4877 static GstRTSPResult
4878 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
4879 gint orig_rtpport, gint orig_rtcpport)
4882 gint nr_udp, nr_int;
4884 gint rtpport = 0, rtcpport = 0;
4887 src = stream->parent;
4889 /* find number of placeholders first */
4890 if (strstr (*transports, "%%i2"))
4892 else if (strstr (*transports, "%%i1"))
4897 if (strstr (*transports, "%%u2"))
4899 else if (strstr (*transports, "%%u1"))
4904 if (nr_udp == 0 && nr_int == 0)
4908 if (!orig_rtpport || !orig_rtcpport) {
4909 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
4912 rtpport = orig_rtpport;
4913 rtcpport = orig_rtcpport;
4917 str = g_string_new ("");
4919 while ((next = strstr (p, "%%"))) {
4920 g_string_append_len (str, p, next - p);
4921 if (next[2] == 'u') {
4923 g_string_append_printf (str, "%d", rtpport);
4924 else if (next[3] == '2')
4925 g_string_append_printf (str, "%d", rtcpport);
4927 if (next[2] == 'i') {
4929 g_string_append_printf (str, "%d", src->free_channel);
4930 else if (next[3] == '2')
4931 g_string_append_printf (str, "%d", src->free_channel + 1);
4936 /* append final part */
4937 g_string_append (str, p);
4939 g_free (*transports);
4940 *transports = g_string_free (str, FALSE);
4948 return GST_RTSP_ERROR;
4953 gst_rtspsrc_stream_is_real_media (GstRTSPStream * stream)
4955 gboolean res = FALSE;
4959 const gchar *enc = NULL;
4961 s = gst_caps_get_structure (stream->caps, 0);
4962 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
4963 res = (strstr (enc, "-REAL") != NULL);
4969 /* Perform the SETUP request for all the streams.
4971 * We ask the server for a specific transport, which initially includes all the
4972 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
4973 * two local UDP ports that we send to the server.
4975 * Once the server replied with a transport, we configure the other streams
4976 * with the same transport.
4978 * This function will also configure the stream for the selected transport,
4979 * which basically means creating the pipeline.
4981 static GstRTSPResult
4982 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
4985 GstRTSPResult res = GST_RTSP_ERROR;
4986 GstRTSPMessage request = { 0 };
4987 GstRTSPMessage response = { 0 };
4988 GstRTSPStream *stream = NULL;
4989 GstRTSPLowerTrans protocols;
4990 GstRTSPStatusCode code;
4991 gboolean unsupported_real = FALSE;
4992 gint rtpport, rtcpport;
4996 if (src->conninfo.connection) {
4997 url = gst_rtsp_connection_get_url (src->conninfo.connection);
4998 /* we initially allow all configured lower transports. based on the URL
4999 * transports and the replies from the server we narrow them down. */
5000 protocols = url->transports & src->cur_protocols;
5003 protocols = src->cur_protocols;
5009 /* reset some state */
5010 src->free_channel = 0;
5011 src->interleaved = FALSE;
5012 src->need_activate = FALSE;
5013 /* keep track of next port number, 0 is random */
5014 src->next_port_num = src->client_port_range.min;
5015 rtpport = rtcpport = 0;
5017 if (G_UNLIKELY (src->streams == NULL))
5020 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5021 GstRTSPConnection *conn;
5026 stream = (GstRTSPStream *) walk->data;
5028 /* see if we need to configure this stream */
5029 if (!gst_rtsp_ext_list_configure_stream (src->extensions, stream->caps)) {
5030 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
5032 stream->disabled = TRUE;
5036 /* merge/overwrite global caps */
5041 s = gst_caps_get_structure (stream->caps, 0);
5043 num = gst_structure_n_fields (src->props);
5044 for (j = 0; j < num; j++) {
5048 name = gst_structure_nth_field_name (src->props, j);
5049 val = gst_structure_get_value (src->props, name);
5050 gst_structure_set_value (s, name, val);
5052 GST_DEBUG_OBJECT (src, "copied %s", name);
5056 /* skip setup if we have no URL for it */
5057 if (stream->conninfo.location == NULL) {
5058 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
5062 if (src->conninfo.connection == NULL) {
5063 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
5064 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
5067 conn = stream->conninfo.connection;
5069 conn = src->conninfo.connection;
5071 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
5072 stream->conninfo.location);
5074 /* if we have a multicast connection, only suggest multicast from now on */
5075 if (stream->is_multicast)
5076 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
5079 /* first selectable protocol */
5080 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5082 if (!protocol_masks[mask])
5086 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
5087 protocol_masks[mask]);
5088 /* create a string with first transport in line */
5090 res = gst_rtspsrc_create_transports_string (src,
5091 protocols & protocol_masks[mask], &transports);
5092 if (res < 0 || transports == NULL)
5093 goto setup_transport_failed;
5095 if (strlen (transports) == 0) {
5096 g_free (transports);
5097 GST_DEBUG_OBJECT (src, "no transports found");
5102 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
5104 /* replace placeholders with real values, this function will optionally
5105 * allocate UDP ports and other info needed to execute the setup request */
5106 res = gst_rtspsrc_prepare_transports (stream, &transports,
5107 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
5109 g_free (transports);
5110 goto setup_transport_failed;
5113 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
5115 /* create SETUP request */
5117 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
5118 stream->conninfo.location);
5120 g_free (transports);
5121 goto create_request_failed;
5124 /* select transport, copy is made when adding to header so we can free it. */
5125 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
5126 g_free (transports);
5128 /* if the user wants a non default RTP packet size we add the blocksize
5130 if (src->rtp_blocksize > 0) {
5131 hval = g_strdup_printf ("%d", src->rtp_blocksize);
5132 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
5137 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
5140 /* handle the code ourselves */
5141 if ((res = gst_rtspsrc_send (src, conn, &request, &response, &code) < 0))
5145 case GST_RTSP_STS_OK:
5147 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
5148 gst_rtsp_message_unset (&request);
5149 gst_rtsp_message_unset (&response);
5150 /* cleanup of leftover transport */
5151 gst_rtspsrc_stream_free_udp (stream);
5152 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
5153 * we might be in this case */
5154 if (stream->container && rtpport && rtcpport && !retry) {
5155 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
5160 /* this transport did not go down well, but we may have others to try
5161 * that we did not send yet, try those and only give up then
5162 * but not without checking for lost cause/extension so we can
5163 * post a nicer/more useful error message later */
5164 if (!unsupported_real)
5165 unsupported_real = gst_rtspsrc_stream_is_real_media (stream);
5166 /* select next available protocol, give up on this stream if none */
5168 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
5170 if (!protocol_masks[mask] || unsupported_real)
5175 /* cleanup of leftover transport and move to the next stream */
5176 gst_rtspsrc_stream_free_udp (stream);
5177 goto response_error;
5180 /* parse response transport */
5182 gchar *resptrans = NULL;
5183 GstRTSPTransport transport = { 0 };
5185 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
5188 gst_rtspsrc_stream_free_udp (stream);
5192 /* parse transport, go to next stream on parse error */
5193 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
5194 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
5198 /* update allowed transports for other streams. once the transport of
5199 * one stream has been determined, we make sure that all other streams
5200 * are configured in the same way */
5201 switch (transport.lower_transport) {
5202 case GST_RTSP_LOWER_TRANS_TCP:
5203 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
5204 protocols = GST_RTSP_LOWER_TRANS_TCP;
5205 src->interleaved = TRUE;
5206 /* update free channels */
5208 MAX (transport.interleaved.min, src->free_channel);
5210 MAX (transport.interleaved.max, src->free_channel);
5211 src->free_channel++;
5213 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
5214 /* only allow multicast for other streams */
5215 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
5216 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
5218 case GST_RTSP_LOWER_TRANS_UDP:
5219 /* only allow unicast for other streams */
5220 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
5221 protocols = GST_RTSP_LOWER_TRANS_UDP;
5224 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
5225 transport.lower_transport);
5229 if (!stream->container || (!src->interleaved && !retry)) {
5230 /* now configure the stream with the selected transport */
5231 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
5232 GST_DEBUG_OBJECT (src,
5233 "could not configure stream %p transport, skipping stream",
5236 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
5237 /* retain the first allocated UDP port pair */
5238 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
5239 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
5242 /* we need to activate at least one streams when we detect activity */
5243 src->need_activate = TRUE;
5245 /* clean up our transport struct */
5246 gst_rtsp_transport_init (&transport);
5247 /* clean up used RTSP messages */
5248 gst_rtsp_message_unset (&request);
5249 gst_rtsp_message_unset (&response);
5253 /* store the transport protocol that was configured */
5254 src->cur_protocols = protocols;
5256 gst_rtsp_ext_list_stream_select (src->extensions, url);
5258 /* if there is nothing to activate, error out */
5259 if (!src->need_activate)
5260 goto nothing_to_activate;
5267 /* no transport possible, post an error and stop */
5268 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5269 ("Could not connect to server, no protocols left"));
5270 return GST_RTSP_ERROR;
5274 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5275 ("SDP contains no streams"));
5276 return GST_RTSP_ERROR;
5278 create_request_failed:
5280 gchar *str = gst_rtsp_strresult (res);
5282 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5283 ("Could not create request. (%s)", str));
5287 setup_transport_failed:
5289 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5290 ("Could not setup transport."));
5291 res = GST_RTSP_ERROR;
5296 const gchar *str = gst_rtsp_status_as_text (code);
5298 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5299 ("Error (%d): %s", code, GST_STR_NULL (str)));
5300 res = GST_RTSP_ERROR;
5305 gchar *str = gst_rtsp_strresult (res);
5307 if (res != GST_RTSP_EINTR) {
5308 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5309 ("Could not send message. (%s)", str));
5311 GST_WARNING_OBJECT (src, "send interrupted");
5318 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5319 ("Server did not select transport."));
5320 res = GST_RTSP_ERROR;
5323 nothing_to_activate:
5325 /* none of the available error codes is really right .. */
5326 if (unsupported_real) {
5327 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5328 (_("No supported stream was found. You might need to install a "
5329 "GStreamer RTSP extension plugin for Real media streams.")),
5332 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
5333 (_("No supported stream was found. You might need to allow "
5334 "more transport protocols or may otherwise be missing "
5335 "the right GStreamer RTSP extension plugin.")), (NULL));
5337 return GST_RTSP_ERROR;
5341 gst_rtsp_message_unset (&request);
5342 gst_rtsp_message_unset (&response);
5348 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
5349 GstSegment * segment)
5352 GstRTSPTimeRange *therange;
5355 gst_rtsp_range_free (src->range);
5357 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
5358 GST_DEBUG_OBJECT (src, "parsed range %s", range);
5359 src->range = therange;
5361 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
5363 gst_segment_init (segment, GST_FORMAT_TIME);
5367 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
5368 therange->min.type, therange->min.seconds, therange->max.type,
5369 therange->max.seconds);
5371 if (therange->min.type == GST_RTSP_TIME_NOW)
5373 else if (therange->min.type == GST_RTSP_TIME_END)
5376 seconds = therange->min.seconds * GST_SECOND;
5378 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
5379 GST_TIME_ARGS (seconds));
5381 /* we need to start playback without clipping from the position reported by
5383 segment->start = seconds;
5384 segment->position = seconds;
5386 if (therange->max.type == GST_RTSP_TIME_NOW)
5388 else if (therange->max.type == GST_RTSP_TIME_END)
5391 seconds = therange->max.seconds * GST_SECOND;
5393 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
5394 GST_TIME_ARGS (seconds));
5396 /* live (WMS) server might send overflowed large max as its idea of infinity,
5397 * compensate to prevent problems later on */
5398 if (seconds != -1 && seconds < 0) {
5400 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
5403 /* live (WMS) might send min == max, which is not worth recording */
5404 if (segment->duration == -1 && seconds == segment->start)
5407 /* don't change duration with unknown value, we might have a valid value
5408 * there that we want to keep. */
5410 segment->duration = seconds;
5415 /* must be called with the RTSP state lock */
5416 static GstRTSPResult
5417 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
5423 /* prepare global stream caps properties */
5425 gst_structure_remove_all_fields (src->props);
5427 src->props = gst_structure_new_empty ("RTSPProperties");
5430 gst_sdp_message_dump (sdp);
5432 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
5434 gst_segment_init (&src->segment, GST_FORMAT_TIME);
5436 /* parse range for duration reporting. */
5441 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
5445 /* keep track of the range and configure it in the segment */
5446 if (gst_rtspsrc_parse_range (src, range, &src->segment))
5450 /* try to find a global control attribute. Note that a '*' means that we should
5451 * do aggregate control with the current url (so we don't do anything and
5452 * leave the current connection as is) */
5454 const gchar *control;
5457 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
5458 if (control == NULL)
5461 /* only take fully qualified urls */
5462 if (g_str_has_prefix (control, "rtsp://"))
5466 g_free (src->conninfo.location);
5467 src->conninfo.location = g_strdup (control);
5468 /* make a connection for this, if there was a connection already, nothing
5470 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
5471 GST_ERROR_OBJECT (src, "could not connect");
5474 /* we need to keep the control url separate from the connection url because
5475 * the rules for constructing the media control url need it */
5476 g_free (src->control);
5477 src->control = g_strdup (control);
5480 /* create streams */
5481 n_streams = gst_sdp_message_medias_len (sdp);
5482 for (i = 0; i < n_streams; i++) {
5483 gst_rtspsrc_create_stream (src, sdp, i);
5486 src->state = GST_RTSP_STATE_INIT;
5489 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
5492 /* reset our state */
5493 src->need_range = TRUE;
5496 src->state = GST_RTSP_STATE_READY;
5503 GST_ERROR_OBJECT (src, "setup failed");
5508 static GstRTSPResult
5509 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
5513 GstRTSPMessage request = { 0 };
5514 GstRTSPMessage response = { 0 };
5517 gchar *respcont = NULL;
5520 src->need_redirect = FALSE;
5522 /* can't continue without a valid url */
5523 if (G_UNLIKELY (src->conninfo.url == NULL)) {
5524 res = GST_RTSP_EINVAL;
5527 src->tried_url_auth = FALSE;
5529 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
5530 goto connect_failed;
5532 /* create OPTIONS */
5533 GST_DEBUG_OBJECT (src, "create options...");
5535 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
5536 src->conninfo.url_str);
5538 goto create_request_failed;
5541 GST_DEBUG_OBJECT (src, "send options...");
5544 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
5547 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5552 if (!gst_rtspsrc_parse_methods (src, &response))
5555 /* create DESCRIBE */
5556 GST_DEBUG_OBJECT (src, "create describe...");
5558 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
5559 src->conninfo.url_str);
5561 goto create_request_failed;
5563 /* we only accept SDP for now */
5564 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
5568 GST_DEBUG_OBJECT (src, "send describe...");
5571 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
5574 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
5578 /* we only perform redirect for the describe, currently */
5579 if (src->need_redirect) {
5580 /* close connection, we don't have to send a TEARDOWN yet, ignore the
5582 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5584 gst_rtsp_message_unset (&request);
5585 gst_rtsp_message_unset (&response);
5591 /* it could be that the DESCRIBE method was not implemented */
5592 if (!src->methods & GST_RTSP_DESCRIBE)
5595 /* check if reply is SDP */
5596 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
5598 /* could not be set but since the request returned OK, we assume it
5599 * was SDP, else check it. */
5601 if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
5602 goto wrong_content_type;
5605 /* get message body and parse as SDP */
5606 gst_rtsp_message_get_body (&response, &data, &size);
5607 if (data == NULL || size == 0)
5610 GST_DEBUG_OBJECT (src, "parse SDP...");
5611 gst_sdp_message_new (sdp);
5612 gst_sdp_message_parse_buffer (data, size, *sdp);
5614 /* clean up any messages */
5615 gst_rtsp_message_unset (&request);
5616 gst_rtsp_message_unset (&response);
5623 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
5624 ("No valid RTSP URL was provided"));
5629 gchar *str = gst_rtsp_strresult (res);
5631 if (res != GST_RTSP_EINTR) {
5632 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
5633 ("Failed to connect. (%s)", str));
5635 GST_WARNING_OBJECT (src, "connect interrupted");
5640 create_request_failed:
5642 gchar *str = gst_rtsp_strresult (res);
5644 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5645 ("Could not create request. (%s)", str));
5651 /* Don't post a message - the rtsp_send method will have
5652 * taken care of it because we passed NULL for the response code */
5657 /* error was posted */
5658 res = GST_RTSP_ERROR;
5663 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5664 ("Server does not support SDP, got %s.", respcont));
5665 res = GST_RTSP_ERROR;
5670 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
5671 ("Server can not provide an SDP."));
5672 res = GST_RTSP_ERROR;
5677 if (src->conninfo.connection) {
5678 GST_DEBUG_OBJECT (src, "free connection");
5679 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5681 gst_rtsp_message_unset (&request);
5682 gst_rtsp_message_unset (&response);
5687 static GstRTSPResult
5688 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
5693 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
5695 if (src->sdp == NULL) {
5696 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
5700 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
5705 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
5712 GST_WARNING_OBJECT (src, "can't get sdp");
5713 src->open_error = TRUE;
5718 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
5719 src->open_error = TRUE;
5724 static GstRTSPResult
5725 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
5727 GstRTSPMessage request = { 0 };
5728 GstRTSPMessage response = { 0 };
5729 GstRTSPResult res = GST_RTSP_OK;
5733 GST_DEBUG_OBJECT (src, "TEARDOWN...");
5735 if (src->state < GST_RTSP_STATE_READY) {
5736 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
5743 /* construct a control url */
5745 control = src->control;
5747 control = src->conninfo.url_str;
5749 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
5752 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5753 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5755 GstRTSPConnInfo *info;
5757 /* try aggregate control first but do non-aggregate control otherwise */
5759 setup_url = control;
5760 else if ((setup_url = stream->conninfo.location) == NULL)
5763 if (src->conninfo.connection) {
5764 info = &src->conninfo;
5765 } else if (stream->conninfo.connection) {
5766 info = &stream->conninfo;
5770 if (!info->connected)
5775 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
5777 goto create_request_failed;
5780 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
5783 gst_rtspsrc_send (src, info->connection, &request, &response,
5787 /* FIXME, parse result? */
5788 gst_rtsp_message_unset (&request);
5789 gst_rtsp_message_unset (&response);
5792 /* early exit when we did aggregate control */
5798 /* close connections */
5799 GST_DEBUG_OBJECT (src, "closing connection...");
5800 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
5801 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5802 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5803 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
5807 gst_rtspsrc_cleanup (src);
5809 src->state = GST_RTSP_STATE_INVALID;
5812 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
5817 create_request_failed:
5819 gchar *str = gst_rtsp_strresult (res);
5821 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
5822 ("Could not create request. (%s)", str));
5828 gchar *str = gst_rtsp_strresult (res);
5830 gst_rtsp_message_unset (&request);
5831 if (res != GST_RTSP_EINTR) {
5832 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5833 ("Could not send message. (%s)", str));
5835 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
5842 GST_DEBUG_OBJECT (src,
5843 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
5848 /* RTP-Info is of the format:
5850 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
5852 * rtptime corresponds to the timestamp for the NPT time given in the header
5853 * seqbase corresponds to the next sequence number we received. This number
5854 * indicates the first seqnum after the seek and should be used to discard
5855 * packets that are from before the seek.
5858 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
5863 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
5865 infos = g_strsplit (rtpinfo, ",", 0);
5866 for (i = 0; infos[i]; i++) {
5868 GstRTSPStream *stream;
5872 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
5874 /* init values, types of seqbase and timebase are bigger than needed so we
5875 * can store -1 as uninitialized values */
5880 /* parse url, find stream for url.
5881 * parse seq and rtptime. The seq number should be configured in the rtp
5882 * depayloader or session manager to detect gaps. Same for the rtptime, it
5883 * should be used to create an initial time newsegment. */
5884 fields = g_strsplit (infos[i], ";", 0);
5885 for (j = 0; fields[j]; j++) {
5886 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
5887 /* remove leading whitespace */
5888 fields[j] = g_strchug (fields[j]);
5889 if (g_str_has_prefix (fields[j], "url=")) {
5890 /* get the url and the stream */
5892 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
5893 } else if (g_str_has_prefix (fields[j], "seq=")) {
5894 seqbase = atoi (fields[j] + 4);
5895 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
5896 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
5899 g_strfreev (fields);
5900 /* now we need to store the values for the caps of the stream */
5901 if (stream != NULL) {
5902 GST_DEBUG_OBJECT (src,
5903 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
5904 stream, seqbase, timebase);
5906 /* we have a stream, configure detected params */
5907 stream->seqbase = seqbase;
5908 stream->timebase = timebase;
5917 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
5922 interval = strtoul (rtcp, NULL, 10);
5923 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
5928 interval *= GST_MSECOND;
5930 for (walk = src->streams; walk; walk = g_list_next (walk)) {
5931 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
5933 /* already (optionally) retrieved this when configuring manager */
5934 if (stream->session) {
5935 GObject *rtpsession = stream->session;
5937 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
5939 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
5943 /* now it happens that (Xenon) server sending this may also provide bogus
5944 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
5945 * and just use RTP-Info to sync */
5947 GObjectClass *klass;
5949 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
5950 if (g_object_class_find_property (klass, "rtcp-sync")) {
5951 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
5952 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
5958 gst_rtspsrc_get_float (const gchar * dstr)
5960 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
5962 /* canonicalise floating point string so we can handle float strings
5963 * in the form "24.930" or "24,930" irrespective of the current locale */
5964 g_strlcpy (s, dstr, sizeof (s));
5965 g_strdelimit (s, ",", '.');
5966 return g_ascii_strtod (s, NULL);
5970 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
5972 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
5974 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
5975 g_strlcpy (val_str, "now", sizeof (val_str));
5977 if (segment->position == 0) {
5978 g_strlcpy (val_str, "0", sizeof (val_str));
5980 g_ascii_dtostr (val_str, sizeof (val_str),
5981 ((gdouble) segment->position) / GST_SECOND);
5984 return g_strdup_printf ("npt=%s-", val_str);
5988 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
5990 stream->timebase = -1;
5991 stream->seqbase = -1;
5995 stream->caps = gst_caps_make_writable (stream->caps);
5996 s = gst_caps_get_structure (stream->caps, 0);
5997 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
6001 static GstRTSPResult
6002 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
6004 GstRTSPResult res = GST_RTSP_OK;
6006 if (src->state < GST_RTSP_STATE_READY) {
6007 res = GST_RTSP_ERROR;
6008 if (src->open_error) {
6009 GST_DEBUG_OBJECT (src, "the stream was in error");
6013 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
6015 if ((res = gst_rtspsrc_open (src, async)) < 0) {
6016 GST_DEBUG_OBJECT (src, "failed to open stream");
6025 static GstRTSPResult
6026 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
6028 GstRTSPMessage request = { 0 };
6029 GstRTSPMessage response = { 0 };
6030 GstRTSPResult res = GST_RTSP_OK;
6036 GST_DEBUG_OBJECT (src, "PLAY...");
6038 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6041 if (!(src->methods & GST_RTSP_PLAY))
6044 if (src->state == GST_RTSP_STATE_PLAYING)
6047 if (!src->conninfo.connection || !src->conninfo.connected)
6050 /* send some dummy packets before we activate the receive in the
6052 gst_rtspsrc_send_dummy_packets (src);
6054 /* activate receive elements;
6055 * only in async case, since receive elements may not have been affected
6056 * by overall state change (e.g. not around yet),
6057 * do not mess with state in sync case (e.g. seeking) */
6059 gst_element_set_state (GST_ELEMENT_CAST (src), GST_STATE_PLAYING);
6061 /* construct a control url */
6063 control = src->control;
6065 control = src->conninfo.url_str;
6067 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6068 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6070 GstRTSPConnection *conn;
6072 /* try aggregate control first but do non-aggregate control otherwise */
6074 setup_url = control;
6075 else if ((setup_url = stream->conninfo.location) == NULL)
6078 if (src->conninfo.connection) {
6079 conn = src->conninfo.connection;
6080 } else if (stream->conninfo.connection) {
6081 conn = stream->conninfo.connection;
6087 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
6089 goto create_request_failed;
6091 if (src->need_range) {
6092 hval = gen_range_header (src, segment);
6094 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
6098 if (segment->rate != 1.0) {
6099 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
6101 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
6103 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
6105 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
6109 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
6111 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6114 /* seek may have silently failed as it is not supported */
6115 if (!(src->methods & GST_RTSP_PLAY)) {
6116 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
6117 /* obviously it is supported as we made it here */
6118 src->methods |= GST_RTSP_PLAY;
6119 src->seekable = FALSE;
6120 /* but there is nothing to parse in the response,
6121 * so convey we have no idea and not to expect anything particular */
6122 clear_rtp_base (src, stream);
6126 /* need to do for all streams */
6127 for (run = src->streams; run; run = g_list_next (run))
6128 clear_rtp_base (src, (GstRTSPStream *) run->data);
6130 /* NOTE the above also disables npt based eos detection */
6131 /* and below forces position to 0,
6132 * which is visible feedback we lost the plot */
6133 segment->start = segment->position = src->last_pos;
6136 gst_rtsp_message_unset (&request);
6138 /* parse RTP npt field. This is the current position in the stream (Normal
6139 * Play Time) and should be put in the NEWSEGMENT position field. */
6140 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
6142 gst_rtspsrc_parse_range (src, hval, segment);
6144 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
6145 segment->rate = 1.0;
6147 /* parse Speed header. This is the intended playback rate of the stream
6148 * and should be put in the NEWSEGMENT rate field. */
6149 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
6150 0) == GST_RTSP_OK) {
6151 segment->rate = gst_rtspsrc_get_float (hval);
6152 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
6153 &hval, 0) == GST_RTSP_OK) {
6154 segment->rate = gst_rtspsrc_get_float (hval);
6157 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
6158 * for the RTP packets. If this is not present, we assume all starts from 0...
6159 * This is info for the RTP session manager that we pass to it in caps. */
6161 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
6162 &hval, hval_idx++) == GST_RTSP_OK)
6163 gst_rtspsrc_parse_rtpinfo (src, hval);
6165 /* some servers indicate RTCP parameters in PLAY response,
6166 * rather than properly in SDP */
6167 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
6168 &hval, 0) == GST_RTSP_OK)
6169 gst_rtspsrc_handle_rtcp_interval (src, hval);
6171 gst_rtsp_message_unset (&response);
6173 /* early exit when we did aggregate control */
6177 /* set again when needed */
6178 src->need_range = FALSE;
6180 /* configure the caps of the streams after we parsed all headers. */
6181 gst_rtspsrc_configure_caps (src, segment);
6183 src->running = TRUE;
6184 src->base_time = -1;
6185 src->state = GST_RTSP_STATE_PLAYING;
6188 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
6189 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6190 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6191 stream->discont = TRUE;
6196 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
6203 GST_DEBUG_OBJECT (src, "failed to open stream");
6208 GST_DEBUG_OBJECT (src, "PLAY is not supported");
6213 GST_DEBUG_OBJECT (src, "we were already PLAYING");
6216 create_request_failed:
6218 gchar *str = gst_rtsp_strresult (res);
6220 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6221 ("Could not create request. (%s)", str));
6227 gchar *str = gst_rtsp_strresult (res);
6229 gst_rtsp_message_unset (&request);
6230 if (res != GST_RTSP_EINTR) {
6231 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6232 ("Could not send message. (%s)", str));
6234 GST_WARNING_OBJECT (src, "PLAY interrupted");
6241 static GstRTSPResult
6242 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle, gboolean async)
6244 GstRTSPResult res = GST_RTSP_OK;
6245 GstRTSPMessage request = { 0 };
6246 GstRTSPMessage response = { 0 };
6250 GST_DEBUG_OBJECT (src, "PAUSE...");
6252 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
6255 if (!(src->methods & GST_RTSP_PAUSE))
6258 if (src->state == GST_RTSP_STATE_READY)
6261 if (!src->conninfo.connection || !src->conninfo.connected)
6264 /* construct a control url */
6266 control = src->control;
6268 control = src->conninfo.url_str;
6270 /* loop over the streams. We might exit the loop early when we could do an
6271 * aggregate control */
6272 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6273 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
6274 GstRTSPConnection *conn;
6277 /* try aggregate control first but do non-aggregate control otherwise */
6279 setup_url = control;
6280 else if ((setup_url = stream->conninfo.location) == NULL)
6283 if (src->conninfo.connection) {
6284 conn = src->conninfo.connection;
6285 } else if (stream->conninfo.connection) {
6286 conn = stream->conninfo.connection;
6292 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
6293 ("Sending PAUSE request"));
6296 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
6298 goto create_request_failed;
6300 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
6303 gst_rtsp_message_unset (&request);
6304 gst_rtsp_message_unset (&response);
6306 /* exit early when we did agregate control */
6312 src->state = GST_RTSP_STATE_READY;
6316 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
6323 GST_DEBUG_OBJECT (src, "failed to open stream");
6328 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
6333 GST_DEBUG_OBJECT (src, "we were already PAUSED");
6336 create_request_failed:
6338 gchar *str = gst_rtsp_strresult (res);
6340 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6341 ("Could not create request. (%s)", str));
6347 gchar *str = gst_rtsp_strresult (res);
6349 gst_rtsp_message_unset (&request);
6350 if (res != GST_RTSP_EINTR) {
6351 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6352 ("Could not send message. (%s)", str));
6354 GST_WARNING_OBJECT (src, "PAUSE interrupted");
6362 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
6364 GstRTSPSrc *rtspsrc;
6366 rtspsrc = GST_RTSPSRC (bin);
6368 switch (GST_MESSAGE_TYPE (message)) {
6369 case GST_MESSAGE_EOS:
6370 gst_message_unref (message);
6372 case GST_MESSAGE_ELEMENT:
6374 const GstStructure *s = gst_message_get_structure (message);
6376 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
6377 gboolean ignore_timeout;
6379 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
6381 GST_OBJECT_LOCK (rtspsrc);
6382 ignore_timeout = rtspsrc->ignore_timeout;
6383 rtspsrc->ignore_timeout = TRUE;
6384 GST_OBJECT_UNLOCK (rtspsrc);
6386 /* we only act on the first udp timeout message, others are irrelevant
6387 * and can be ignored. */
6388 if (!ignore_timeout)
6389 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, TRUE);
6391 gst_message_unref (message);
6394 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6397 case GST_MESSAGE_ERROR:
6400 GstRTSPStream *stream;
6403 udpsrc = GST_MESSAGE_SRC (message);
6405 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
6406 GST_ELEMENT_NAME (udpsrc));
6408 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
6412 /* we ignore the RTCP udpsrc */
6413 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
6416 /* if we get error messages from the udp sources, that's not a problem as
6417 * long as not all of them error out. We also don't really know what the
6418 * problem is, the message does not give enough detail... */
6419 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
6420 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
6421 if (ret != GST_FLOW_OK)
6425 gst_message_unref (message);
6429 /* fatal but not our message, forward */
6430 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6435 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
6441 /* the thread where everything happens */
6443 gst_rtspsrc_thread (GstRTSPSrc * src)
6447 gboolean running = FALSE;
6449 GST_OBJECT_LOCK (src);
6450 cmd = src->loop_cmd;
6451 src->loop_cmd = CMD_WAIT;
6452 GST_DEBUG_OBJECT (src, "got command %d", cmd);
6454 /* we got the message command, so ensure communication is possible again */
6455 gst_rtspsrc_connection_flush (src, FALSE);
6457 /* we allow these to be interrupted */
6458 if (cmd == CMD_LOOP || cmd == CMD_CLOSE || cmd == CMD_PAUSE)
6459 src->waiting = TRUE;
6460 GST_OBJECT_UNLOCK (src);
6464 ret = gst_rtspsrc_open (src, TRUE);
6467 ret = gst_rtspsrc_play (src, &src->segment, TRUE);
6468 if (ret == GST_RTSP_OK)
6472 ret = gst_rtspsrc_pause (src, TRUE, TRUE);
6473 if (ret == GST_RTSP_OK)
6477 ret = gst_rtspsrc_close (src, TRUE, FALSE);
6480 running = gst_rtspsrc_loop (src);
6483 ret = gst_rtspsrc_reconnect (src, FALSE);
6484 if (ret == GST_RTSP_OK)
6491 GST_OBJECT_LOCK (src);
6492 /* and go back to sleep */
6493 if (src->loop_cmd == CMD_WAIT) {
6495 src->loop_cmd = CMD_LOOP;
6497 gst_task_pause (src->task);
6500 src->waiting = FALSE;
6501 GST_OBJECT_UNLOCK (src);
6505 gst_rtspsrc_start (GstRTSPSrc * src)
6507 GST_DEBUG_OBJECT (src, "starting");
6509 GST_OBJECT_LOCK (src);
6511 src->loop_cmd = CMD_WAIT;
6513 if (src->task == NULL) {
6514 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src);
6515 if (src->task == NULL)
6518 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
6520 GST_OBJECT_UNLOCK (src);
6527 GST_ERROR_OBJECT (src, "failed to create task");
6533 gst_rtspsrc_stop (GstRTSPSrc * src)
6537 GST_DEBUG_OBJECT (src, "stopping");
6539 /* also cancels pending task */
6540 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, TRUE);
6542 GST_OBJECT_LOCK (src);
6543 if ((task = src->task)) {
6545 GST_OBJECT_UNLOCK (src);
6547 gst_task_stop (task);
6549 /* make sure it is not running */
6550 GST_RTSP_STREAM_LOCK (src);
6551 GST_RTSP_STREAM_UNLOCK (src);
6553 /* now wait for the task to finish */
6554 gst_task_join (task);
6556 /* and free the task */
6557 gst_object_unref (GST_OBJECT (task));
6559 GST_OBJECT_LOCK (src);
6561 GST_OBJECT_UNLOCK (src);
6563 /* ensure synchronously all is closed and clean */
6564 gst_rtspsrc_close (src, FALSE, TRUE);
6569 static GstStateChangeReturn
6570 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
6572 GstRTSPSrc *rtspsrc;
6573 GstStateChangeReturn ret;
6575 rtspsrc = GST_RTSPSRC (element);
6577 switch (transition) {
6578 case GST_STATE_CHANGE_NULL_TO_READY:
6579 if (!gst_rtspsrc_start (rtspsrc))
6582 case GST_STATE_CHANGE_READY_TO_PAUSED:
6583 /* init some state */
6584 rtspsrc->cur_protocols = rtspsrc->protocols;
6585 /* first attempt, don't ignore timeouts */
6586 rtspsrc->ignore_timeout = FALSE;
6587 rtspsrc->open_error = FALSE;
6588 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, FALSE);
6590 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6591 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6592 /* unblock the tcp tasks and make the loop waiting */
6593 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, TRUE);
6595 case GST_STATE_CHANGE_PAUSED_TO_READY:
6601 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
6602 if (ret == GST_STATE_CHANGE_FAILURE)
6605 switch (transition) {
6606 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
6607 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, FALSE);
6609 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
6610 /* send pause request and keep the idle task around */
6611 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, FALSE);
6612 ret = GST_STATE_CHANGE_NO_PREROLL;
6614 case GST_STATE_CHANGE_READY_TO_PAUSED:
6615 ret = GST_STATE_CHANGE_NO_PREROLL;
6617 case GST_STATE_CHANGE_PAUSED_TO_READY:
6618 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, FALSE);
6620 case GST_STATE_CHANGE_READY_TO_NULL:
6621 gst_rtspsrc_stop (rtspsrc);
6632 GST_DEBUG_OBJECT (rtspsrc, "start failed");
6633 return GST_STATE_CHANGE_FAILURE;
6638 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
6641 GstRTSPSrc *rtspsrc;
6643 rtspsrc = GST_RTSPSRC (element);
6645 if (GST_EVENT_IS_DOWNSTREAM (event)) {
6646 res = gst_rtspsrc_push_event (rtspsrc, event, TRUE);
6648 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
6655 /*** GSTURIHANDLER INTERFACE *************************************************/
6658 gst_rtspsrc_uri_get_type (GType type)
6663 static const gchar *const *
6664 gst_rtspsrc_uri_get_protocols (GType type)
6666 static const gchar *protocols[] =
6667 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp", NULL };
6673 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
6675 GstRTSPSrc *src = GST_RTSPSRC (handler);
6677 /* FIXME: make thread-safe */
6678 return g_strdup (src->conninfo.location);
6682 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
6687 GstRTSPUrl *newurl = NULL;
6688 GstSDPMessage *sdp = NULL;
6690 src = GST_RTSPSRC (handler);
6692 /* same URI, we're fine */
6693 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
6696 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
6697 if ((res = gst_sdp_message_new (&sdp) < 0))
6700 GST_DEBUG_OBJECT (src, "parsing SDP message");
6701 if ((res = gst_sdp_message_parse_uri (uri, sdp) < 0))
6705 GST_DEBUG_OBJECT (src, "parsing URI");
6706 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
6710 /* if worked, free previous and store new url object along with the original
6712 GST_DEBUG_OBJECT (src, "configuring URI");
6713 g_free (src->conninfo.location);
6714 src->conninfo.location = g_strdup (uri);
6715 gst_rtsp_url_free (src->conninfo.url);
6716 src->conninfo.url = newurl;
6717 g_free (src->conninfo.url_str);
6719 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
6721 src->conninfo.url_str = NULL;
6724 gst_sdp_message_free (src->sdp);
6726 src->from_sdp = sdp != NULL;
6728 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
6729 GST_DEBUG_OBJECT (src, "request uri is: %s",
6730 GST_STR_NULL (src->conninfo.url_str));
6737 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
6742 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", res);
6743 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6744 "Could not create SDP");
6749 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", res,
6750 GST_STR_NULL (uri));
6751 gst_sdp_message_free (sdp);
6752 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6758 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
6759 GST_STR_NULL (uri), res);
6760 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
6761 "Invalid RTSP URI");
6767 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
6769 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
6771 iface->get_type = gst_rtspsrc_uri_get_type;
6772 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
6773 iface->get_uri = gst_rtspsrc_uri_get_uri;
6774 iface->set_uri = gst_rtspsrc_uri_set_uri;